From 0c21fccd97f0ff58e6e9699370a09f6ec8946061 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 13 Sep 2013 10:44:44 +0300 Subject: ALSA: asihpi: a couple array out of bounds issues These ->put() functions are called from snd_ctl_elem_write() with user supplied data. snd_asihpi_tuner_band_put() is missing a limit check and the check in snd_asihpi_clksrc_put() can underflow. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index dc632cdc3870..5f2acd35dcb9 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1913,6 +1913,7 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol, struct snd_card_asihpi *asihpi = snd_kcontrol_chip(kcontrol); */ u32 h_control = kcontrol->private_value; + unsigned int idx; u16 band; u16 tuner_bands[HPI_TUNER_BAND_LAST]; u32 num_bands = 0; @@ -1920,7 +1921,10 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol, num_bands = asihpi_tuner_band_query(kcontrol, tuner_bands, HPI_TUNER_BAND_LAST); - band = tuner_bands[ucontrol->value.enumerated.item[0]]; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= ARRAY_SIZE(tuner_bands)) + idx = ARRAY_SIZE(tuner_bands) - 1; + band = tuner_bands[idx]; hpi_handle_error(hpi_tuner_set_band(h_control, band)); return 1; @@ -2383,7 +2387,8 @@ static int snd_asihpi_clksrc_put(struct snd_kcontrol *kcontrol, struct snd_card_asihpi *asihpi = (struct snd_card_asihpi *)(kcontrol->private_data); struct clk_cache *clkcache = &asihpi->cc; - int change, item; + unsigned int item; + int change; u32 h_control = kcontrol->private_value; change = 1; -- cgit From b71495ee425943afd80b8188e5c273254b3557a2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:14:21 +0530 Subject: ALSA: ctxfi: Staticize local symbols Local symbols used only in this file are made static. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctdaio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 0c00eb4088ef..84f86bf63b8f 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -33,7 +33,7 @@ struct daio_rsc_idx { unsigned short right; }; -struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x00, .right = 0x01}, [LINEO2] = {.left = 0x18, .right = 0x19}, [LINEO3] = {.left = 0x08, .right = 0x09}, @@ -44,7 +44,7 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [SPDIFI1] = {.left = 0x95, .right = 0x9d}, }; -struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x40, .right = 0x41}, [LINEO2] = {.left = 0x60, .right = 0x61}, [LINEO3] = {.left = 0x50, .right = 0x51}, -- cgit From 399ae7254e13de9239d74574f9fc39fb91a536cf Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:14:22 +0530 Subject: ALSA: hda/ca0132: Staticize codec_send_command 'codec_send_command' is used only in this file. Make it static. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 6e9876f27d95..54d14793725a 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -759,7 +759,7 @@ struct ca0132_spec { /* * CA0132 codec access */ -unsigned int codec_send_command(struct hda_codec *codec, hda_nid_t nid, +static unsigned int codec_send_command(struct hda_codec *codec, hda_nid_t nid, unsigned int verb, unsigned int parm, unsigned int *res) { unsigned int response; -- cgit From 79ae92dd24bb673dd5ddafdbc27cbf4a1e419525 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 16:03:15 +0530 Subject: ALSA: au88x0: Remove redundant break 'break' after a return statement is redundant. Remove it. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_synth.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 8bef47311e45..922a84bba2ef 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -219,7 +219,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; - break; case 1: /* param 0 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -227,7 +226,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; - break; case 2: /* param 1 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -235,7 +233,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; - break; case 3: /* param 2 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -243,7 +240,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; - break; case 4: /* param 3 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -251,7 +247,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; - break; case 6: /* mute */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -259,20 +254,17 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; - break; case 0xb: - { /* delay */ - /* - printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", - WT_DELAY(wt,0), (int)val); - */ - hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); - return 0xc; - } - break; + /* delay */ + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_DELAY(wt,0), (int)val); + */ + hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); + return 0xc; /* Global WT block parameters */ case 5: /* sramp */ ecx = WT_SRAMP(wt); @@ -291,7 +283,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, break; default: return 0; - break; } /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); -- cgit From bf69aa906ccddacc9e966794869431374953738a Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 16:18:59 +0530 Subject: ALSA: rme9652: Remove redundant break 'break' after return statement is not necessary. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 3cde55b753e2..2907e68150cb 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3996,7 +3996,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); if (status & HDSPM_tcoLockAes) { @@ -4006,9 +4005,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - - break; - case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); @@ -4018,7 +4014,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) if (status & 0x4000000) return 1; /* Lock */ return 0; /* No signal */ - break; default: break; -- cgit From 175ee39e8f4053f95e1948afd75c74552b3a175c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:11 +0200 Subject: ASoC: Remove support for reg_access_defaults No users of reg_access_defaults are left and new drivers are going to use regmap for this, so support for it can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 24 -------------------- sound/soc/soc-cache.c | 63 --------------------------------------------------- sound/soc/soc-core.c | 9 -------- 3 files changed, 96 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..447278a3b3e6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -403,12 +403,6 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg); int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -541,22 +535,6 @@ int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -/** - * struct snd_soc_reg_access - Describes whether a given register is - * readable, writable or volatile. - * - * @reg: the register number - * @read: whether this register is readable - * @write: whether this register is writable - * @vol: whether this register is volatile - */ -struct snd_soc_reg_access { - u16 reg; - u16 read; - u16 write; - u16 vol; -}; - /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * @@ -760,8 +738,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - short reg_access_size; - const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index e72f55428f0b..eaa898f8d808 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -275,66 +275,3 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); - -static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - unsigned int min, max, index; - - codec_drv = codec->driver; - min = 0; - max = codec_drv->reg_access_size - 1; - do { - index = (min + max) / 2; - if (codec_drv->reg_access_default[index].reg == reg) - return index; - if (codec_drv->reg_access_default[index].reg < reg) - min = index + 1; - else - max = index; - } while (min <= max); - return -1; -} - -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].vol; -} -EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); - -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].read; -} -EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); - -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].write; -} -EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d0561312f3b..f5ec301603d8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4224,15 +4224,6 @@ int snd_soc_register_codec(struct device *dev, } } - if (codec_drv->reg_access_size && codec_drv->reg_access_default) { - if (!codec->volatile_register) - codec->volatile_register = snd_soc_default_volatile_register; - if (!codec->readable_register) - codec->readable_register = snd_soc_default_readable_register; - if (!codec->writable_register) - codec->writable_register = snd_soc_default_writable_register; - } - for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- cgit From 2a1212a8342c469cee240cf69fe3001b898cda8e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:12 +0200 Subject: ASoC: Remove snd_soc_bulk_write_raw() No users of snd_soc_bulk_write_raw() are left and new drivers are going to use regmap directly for this, so the function can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- sound/soc/soc-core.c | 7 ------- sound/soc/soc-io.c | 26 -------------------------- 3 files changed, 36 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 447278a3b3e6..3f7de6f992c0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -686,7 +686,6 @@ struct snd_soc_codec { unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - int (*bulk_write_raw)(struct snd_soc_codec *, unsigned int, const void *, size_t); void *reg_cache; const void *reg_def_copy; const struct snd_soc_cache_ops *cache_ops; @@ -1097,8 +1096,6 @@ struct soc_enum { unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); unsigned int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val); -unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec, - unsigned int reg, const void *data, size_t len); /* device driver data */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f5ec301603d8..4ce02e6777e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2298,13 +2298,6 @@ unsigned int snd_soc_write(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_write); -unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec, - unsigned int reg, const void *data, size_t len) -{ - return codec->bulk_write_raw(codec, reg, data, len); -} -EXPORT_SYMBOL_GPL(snd_soc_bulk_write_raw); - /** * snd_soc_update_bits - update codec register bits * @codec: audio codec diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 122c0c18b9dd..4f11d23f2062 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -65,31 +65,6 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) return val; } -/* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects. Any - * data written through this function will not go through the cache as - * it only handles writing to volatile or out of bounds registers. - * - * This is currently only supported for devices using the regmap API - * wrappers. - */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, - unsigned int reg, - const void *data, size_t len) -{ - /* To ensure that we don't get out of sync with the cache, check - * whether the base register is volatile or if we've directly asked - * to bypass the cache. Out of bounds registers are considered - * volatile. - */ - if (!codec->cache_bypass - && !snd_soc_codec_volatile_register(codec, reg) - && reg < codec->driver->reg_cache_size) - return -EINVAL; - - return regmap_raw_write(codec->control_data, reg, data, len); -} - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -119,7 +94,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, memset(&config, 0, sizeof(config)); codec->write = hw_write; codec->read = hw_read; - codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; config.reg_bits = addr_bits; config.val_bits = data_bits; -- cgit From b012aa619e50d22df0835b64a5dcebc221fb8053 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:13 +0200 Subject: ASoC: Remove reg_def_copy reg_def_copy was introduced in commit 3335ddca ("ASoC: soc-cache: Use reg_def_copy instead of reg_cache_default") to keep a copy of the register defaults around in case the register defaults where placed in the __devinitdata section. With the __devinitdata section gone we effectivly keep the same data around twice. This patch removes reg_def_copy and uses reg_cache_default directly instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - sound/soc/soc-cache.c | 10 ++++++---- sound/soc/soc-core.c | 15 --------------- 3 files changed, 6 insertions(+), 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3f7de6f992c0..62f320f56644 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -687,7 +687,6 @@ struct snd_soc_codec { unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; - const void *reg_def_copy; const struct snd_soc_cache_ops *cache_ops; struct mutex cache_rw_mutex; int val_bytes; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index eaa898f8d808..a7f83c0c62ce 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -78,8 +78,8 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; - if (codec->reg_def_copy) - if (snd_soc_get_cache_val(codec->reg_def_copy, + if (codec_drv->reg_cache_default) + if (snd_soc_get_cache_val(codec_drv->reg_cache_default, i, codec_drv->reg_word_size) == val) continue; @@ -121,8 +121,10 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { - if (codec->reg_def_copy) - codec->reg_cache = kmemdup(codec->reg_def_copy, + const struct snd_soc_codec_driver *codec_drv = codec->driver; + + if (codec_drv->reg_cache_default) + codec->reg_cache = kmemdup(codec_drv->reg_cache_default, codec->reg_size, GFP_KERNEL); else codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4ce02e6777e5..bbe833ab657e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4201,20 +4201,6 @@ int snd_soc_register_codec(struct device *dev, if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; codec->reg_size = reg_size; - /* it is necessary to make a copy of the default register cache - * because in the case of using a compression type that requires - * the default register cache to be marked as the - * kernel might have freed the array by the time we initialize - * the cache. - */ - if (codec_drv->reg_cache_default) { - codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default, - reg_size, GFP_KERNEL); - if (!codec->reg_def_copy) { - ret = -ENOMEM; - goto fail_codec_name; - } - } } for (i = 0; i < num_dai; i++) { @@ -4273,7 +4259,6 @@ found: dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", codec->name); snd_soc_cache_exit(codec); - kfree(codec->reg_def_copy); kfree(codec->name); kfree(codec); } -- cgit From a94ed23436fb28bdcdd66e7fcf68ca5f7967e456 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:14 +0200 Subject: ASoC: Remove 'reg_size' field from snd_soc_codec struct The reg_size field is calculated in snd_soc_register_codec() and then used exactly once in snd_soc_flat_cache_init(). Since it is calculated based on other fields from the codec struct just move the calculation to snd_soc_flat_cache_init() and remove the 'reg_size' field from the codec struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - sound/soc/soc-cache.c | 7 +++++-- sound/soc/soc-core.c | 7 ------- 3 files changed, 5 insertions(+), 10 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 62f320f56644..577212629d0f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -662,7 +662,6 @@ struct snd_soc_codec { struct list_head card_list; int num_dai; enum snd_soc_compress_type compress_type; - size_t reg_size; /* reg_cache_size * reg_word_size */ int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); int (*writable_register)(struct snd_soc_codec *, unsigned int); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a7f83c0c62ce..9542c83d2295 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -122,12 +122,15 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv = codec->driver; + size_t reg_size; + + reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (codec_drv->reg_cache_default) codec->reg_cache = kmemdup(codec_drv->reg_cache_default, - codec->reg_size, GFP_KERNEL); + reg_size, GFP_KERNEL); else - codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL); + codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); if (!codec->reg_cache) return -ENOMEM; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bbe833ab657e..af9648426f4f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4159,7 +4159,6 @@ int snd_soc_register_codec(struct device *dev, struct snd_soc_dai_driver *dai_drv, int num_dai) { - size_t reg_size; struct snd_soc_codec *codec; int ret, i; @@ -4197,12 +4196,6 @@ int snd_soc_register_codec(struct device *dev, codec->num_dai = num_dai; mutex_init(&codec->mutex); - /* allocate CODEC register cache */ - if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - codec->reg_size = reg_size; - } - for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- cgit From f90fb3f778042b0b9f9aa1fd48cb76047a25eac0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:15 +0200 Subject: ASoC: Remove infrastructure for supporting multiple cache types The only cache type left is the flat cache and new other cache types won't be added since new drivers are supposed to use regmap directly for IO and caching. This patch removes the snd_soc_cache_ops indirection that was added to support multiple cache types and modifies the code to always use the flat cache directly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 27 ------- sound/soc/soc-cache.c | 195 +++++++++++++++----------------------------------- sound/soc/soc-core.c | 27 +------ 3 files changed, 58 insertions(+), 191 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 577212629d0f..a72af6327987 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -330,7 +330,6 @@ struct soc_enum; struct snd_soc_jack; struct snd_soc_jack_zone; struct snd_soc_jack_pin; -struct snd_soc_cache_ops; #include #include @@ -348,10 +347,6 @@ enum snd_soc_control_type { SND_SOC_REGMAP, }; -enum snd_soc_compress_type { - SND_SOC_FLAT_COMPRESSION = 1, -}; - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, @@ -635,19 +630,6 @@ struct snd_soc_compr_ops { int (*trigger)(struct snd_compr_stream *); }; -/* SoC cache ops */ -struct snd_soc_cache_ops { - const char *name; - enum snd_soc_compress_type id; - int (*init)(struct snd_soc_codec *codec); - int (*exit)(struct snd_soc_codec *codec); - int (*read)(struct snd_soc_codec *codec, unsigned int reg, - unsigned int *value); - int (*write)(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value); - int (*sync)(struct snd_soc_codec *codec); -}; - /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; @@ -661,7 +643,6 @@ struct snd_soc_codec { struct list_head list; struct list_head card_list; int num_dai; - enum snd_soc_compress_type compress_type; int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); int (*writable_register)(struct snd_soc_codec *, unsigned int); @@ -686,7 +667,6 @@ struct snd_soc_codec { unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; - const struct snd_soc_cache_ops *cache_ops; struct mutex cache_rw_mutex; int val_bytes; @@ -735,7 +715,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - enum snd_soc_compress_type compress_type; /* codec bias level */ int (*set_bias_level)(struct snd_soc_codec *, @@ -917,12 +896,6 @@ struct snd_soc_codec_conf { * associated per device */ const char *name_prefix; - - /* - * set this to the desired compression type if you want to - * override the one supplied in codec->driver->compress_type - */ - enum snd_soc_compress_type compress_type; }; struct snd_soc_aux_dev { diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9542c83d2295..1b6663f45b34 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -11,12 +11,9 @@ * option) any later version. */ -#include -#include #include -#include -#include #include +#include #include @@ -66,66 +63,18 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, return -1; } -static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) -{ - int i; - int ret; - const struct snd_soc_codec_driver *codec_drv; - unsigned int val; - - codec_drv = codec->driver; - for (i = 0; i < codec_drv->reg_cache_size; ++i) { - ret = snd_soc_cache_read(codec, i, &val); - if (ret) - return ret; - if (codec_drv->reg_cache_default) - if (snd_soc_get_cache_val(codec_drv->reg_cache_default, - i, codec_drv->reg_word_size) == val) - continue; - - WARN_ON(!snd_soc_codec_writable_register(codec, i)); - - ret = snd_soc_write(codec, i, val); - if (ret) - return ret; - dev_dbg(codec->dev, "ASoC: Synced register %#x, value = %#x\n", - i, val); - } - return 0; -} - -static int snd_soc_flat_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - snd_soc_set_cache_val(codec->reg_cache, reg, value, - codec->driver->reg_word_size); - return 0; -} - -static int snd_soc_flat_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value) -{ - *value = snd_soc_get_cache_val(codec->reg_cache, reg, - codec->driver->reg_word_size); - return 0; -} - -static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) -{ - if (!codec->reg_cache) - return 0; - kfree(codec->reg_cache); - codec->reg_cache = NULL; - return 0; -} - -static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) +int snd_soc_cache_init(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv = codec->driver; size_t reg_size; reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + mutex_init(&codec->cache_rw_mutex); + + dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n", + codec->name); + if (codec_drv->reg_cache_default) codec->reg_cache = kmemdup(codec_drv->reg_cache_default, reg_size, GFP_KERNEL); @@ -137,60 +86,19 @@ static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) return 0; } -/* an array of all supported compression types */ -static const struct snd_soc_cache_ops cache_types[] = { - /* Flat *must* be the first entry for fallback */ - { - .id = SND_SOC_FLAT_COMPRESSION, - .name = "flat", - .init = snd_soc_flat_cache_init, - .exit = snd_soc_flat_cache_exit, - .read = snd_soc_flat_cache_read, - .write = snd_soc_flat_cache_write, - .sync = snd_soc_flat_cache_sync - }, -}; - -int snd_soc_cache_init(struct snd_soc_codec *codec) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(cache_types); ++i) - if (cache_types[i].id == codec->compress_type) - break; - - /* Fall back to flat compression */ - if (i == ARRAY_SIZE(cache_types)) { - dev_warn(codec->dev, "ASoC: Could not match compress type: %d\n", - codec->compress_type); - i = 0; - } - - mutex_init(&codec->cache_rw_mutex); - codec->cache_ops = &cache_types[i]; - - if (codec->cache_ops->init) { - if (codec->cache_ops->name) - dev_dbg(codec->dev, "ASoC: Initializing %s cache for %s codec\n", - codec->cache_ops->name, codec->name); - return codec->cache_ops->init(codec); - } - return -ENOSYS; -} - /* * NOTE: keep in mind that this function might be called * multiple times. */ int snd_soc_cache_exit(struct snd_soc_codec *codec) { - if (codec->cache_ops && codec->cache_ops->exit) { - if (codec->cache_ops->name) - dev_dbg(codec->dev, "ASoC: Destroying %s cache for %s codec\n", - codec->cache_ops->name, codec->name); - return codec->cache_ops->exit(codec); - } - return -ENOSYS; + dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n", + codec->name); + if (!codec->reg_cache) + return 0; + kfree(codec->reg_cache); + codec->reg_cache = NULL; + return 0; } /** @@ -203,18 +111,15 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value) { - int ret; + if (!value) + return -EINVAL; mutex_lock(&codec->cache_rw_mutex); - - if (value && codec->cache_ops && codec->cache_ops->read) { - ret = codec->cache_ops->read(codec, reg, value); - mutex_unlock(&codec->cache_rw_mutex); - return ret; - } - + *value = snd_soc_get_cache_val(codec->reg_cache, reg, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); - return -ENOSYS; + + return 0; } EXPORT_SYMBOL_GPL(snd_soc_cache_read); @@ -228,20 +133,42 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_read); int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + mutex_lock(&codec->cache_rw_mutex); + snd_soc_set_cache_val(codec->reg_cache, reg, value, + codec->driver->reg_word_size); + mutex_unlock(&codec->cache_rw_mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_cache_write); + +static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) +{ + int i; int ret; + const struct snd_soc_codec_driver *codec_drv; + unsigned int val; - mutex_lock(&codec->cache_rw_mutex); + codec_drv = codec->driver; + for (i = 0; i < codec_drv->reg_cache_size; ++i) { + ret = snd_soc_cache_read(codec, i, &val); + if (ret) + return ret; + if (codec_drv->reg_cache_default) + if (snd_soc_get_cache_val(codec_drv->reg_cache_default, + i, codec_drv->reg_word_size) == val) + continue; - if (codec->cache_ops && codec->cache_ops->write) { - ret = codec->cache_ops->write(codec, reg, value); - mutex_unlock(&codec->cache_rw_mutex); - return ret; - } + WARN_ON(!snd_soc_codec_writable_register(codec, i)); - mutex_unlock(&codec->cache_rw_mutex); - return -ENOSYS; + ret = snd_soc_write(codec, i, val); + if (ret) + return ret; + dev_dbg(codec->dev, "ASoC: Synced register %#x, value = %#x\n", + i, val); + } + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_cache_write); /** * snd_soc_cache_sync: Sync the register cache with the hardware. @@ -254,26 +181,16 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write); */ int snd_soc_cache_sync(struct snd_soc_codec *codec) { + const char *name = "flat"; int ret; - const char *name; - if (!codec->cache_sync) { + if (!codec->cache_sync) return 0; - } - - if (!codec->cache_ops || !codec->cache_ops->sync) - return -ENOSYS; - - if (codec->cache_ops->name) - name = codec->cache_ops->name; - else - name = "unknown"; - if (codec->cache_ops->name) - dev_dbg(codec->dev, "ASoC: Syncing %s cache for %s codec\n", - codec->cache_ops->name, codec->name); + dev_dbg(codec->dev, "ASoC: Syncing cache for %s codec\n", + codec->name); trace_snd_soc_cache_sync(codec, name, "start"); - ret = codec->cache_ops->sync(codec); + ret = snd_soc_flat_cache_sync(codec); if (!ret) codec->cache_sync = 0; trace_snd_soc_cache_sync(codec, name, "end"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index af9648426f4f..16a3930c6375 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1590,17 +1590,13 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) soc_remove_codec(codec); } -static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, - enum snd_soc_compress_type compress_type) +static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) { int ret; if (codec->cache_init) return 0; - /* override the compress_type if necessary */ - if (compress_type && codec->compress_type != compress_type) - codec->compress_type = compress_type; ret = snd_soc_cache_init(codec); if (ret < 0) { dev_err(codec->dev, @@ -1615,8 +1611,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; - struct snd_soc_codec_conf *codec_conf; - enum snd_soc_compress_type compress_type; struct snd_soc_dai_link *dai_link; int ret, i, order, dai_fmt; @@ -1640,19 +1634,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) list_for_each_entry(codec, &codec_list, list) { if (codec->cache_init) continue; - /* by default we don't override the compress_type */ - compress_type = 0; - /* check to see if we need to override the compress_type */ - for (i = 0; i < card->num_configs; ++i) { - codec_conf = &card->codec_conf[i]; - if (!strcmp(codec->name, codec_conf->dev_name)) { - compress_type = codec_conf->compress_type; - if (compress_type && compress_type - != codec->compress_type) - break; - } - } - ret = snd_soc_init_codec_cache(codec, compress_type); + ret = snd_soc_init_codec_cache(codec); if (ret < 0) goto base_error; } @@ -4175,11 +4157,6 @@ int snd_soc_register_codec(struct device *dev, goto fail_codec; } - if (codec_drv->compress_type) - codec->compress_type = codec_drv->compress_type; - else - codec->compress_type = SND_SOC_FLAT_COMPRESSION; - codec->write = codec_drv->write; codec->read = codec_drv->read; codec->volatile_register = codec_drv->volatile_register; -- cgit From 56fb7421d23c63cf22ac885d2db2302cefc9e1f1 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 13 Sep 2013 18:09:46 +0100 Subject: ASoC: trace: Make sure trace header doesnt depend on any headers Fix build so that asoc trace event header doesn't depend on other headers. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/trace/events/asoc.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 5fc2dcdd21cd..03996b2bb04f 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -14,6 +14,7 @@ struct snd_soc_codec; struct snd_soc_platform; struct snd_soc_card; struct snd_soc_dapm_widget; +struct snd_soc_dapm_path; /* * Log register events -- cgit From 7f05cc98bd9b10b9a0173f3f5d20c2223fdf7470 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 13 Sep 2013 18:09:47 +0100 Subject: ASoC: core utils: Dont set DMA params for BE substreams BE substreams dont require dummy DMA configs so dont set any. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 29b211e9c060..5e633659c1b3 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -75,7 +75,11 @@ static const struct snd_pcm_hardware dummy_dma_hardware = { static int dummy_dma_open(struct snd_pcm_substream *substream) { - snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* BE's dont need dummy params */ + if (!rtd->dai_link->no_pcm) + snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); return 0; } -- cgit From 8ecb5344fd6409c500c9d5757c3a7130d3d7db5b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:51:38 +0100 Subject: ASoC: cq93vc: Don't use control data for core driver data The platform data is being used to obtain the core driver data for the device (which is a bit of an abuse but not the issue at hand) so reference it directly in order to support refactoring to use regmap. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 23316c887b19..e2c4c0a896e2 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -79,7 +79,7 @@ static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - struct davinci_vc *davinci_vc = codec->control_data; + struct davinci_vc *davinci_vc = codec->dev->platform_data; switch (freq) { case 22579200: -- cgit From a851a2bb2d746ccdec0c7cc6ed1c9774921e721e Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:22:17 +0530 Subject: ASoC: fsl_ssi: Staticize local symbols Local symbols used only in this file are made static. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6b743978d5e..4973be774956 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,7 +748,7 @@ static void fsl_ssi_ac97_init(void) fsl_ssi_setup(fsl_ac97_data); } -void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, +static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; @@ -770,7 +770,7 @@ void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); } -unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, +static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; -- cgit From b51600c01979ab1d1c4df17e8910696547ffb9a2 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sat, 31 Aug 2013 13:43:36 +0100 Subject: ASoC: kirkwood-dma: remove IEC958_SUBFRAME formats The Audio block does not support IEC958 subframes as formatted by ALSA: they're very close, but not close enough. The formats differ by: 3 2 2 2 1 1 1 8 4 0 6 2 8 4 0 PCUVDDDDDDDDDDDDDDDD....AAAATTTT - IEC958 subframe PCUV0000........DDDDDDDDDDDDDDDD - Audio block format Where P = parity, C = channel status, U = user data, V = validity, D = sample data, A = aux, T = preamble. As can be seen, the position of the sample is in a different position, and the audio block does not have the aux or preamble bits. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b238434f92b0..0c85c4e1a1ae 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -29,9 +29,7 @@ #define KIRKWOOD_FORMATS \ (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE | \ - SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | \ - SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE) + SNDRV_PCM_FMTBIT_S32_LE) static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) { -- cgit From 38f2b8cbfb1ef517af8af5a63bdff073b7b078fd Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 2 Sep 2013 17:56:18 -0300 Subject: ASoC: mxs: mxs-sgtl5000: Simplify probe function mxs is a device tree only platform, which allows us to simplify a bit mxs_sgtl5000_probe(), because there is no need to check whether device tree is supported or not. Remove mxs_sgtl5000_probe_dt() and place its content inside mxs_sgtl5000_probe() for making the code simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 4bb273786ff3..61822cc53bd3 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -122,14 +122,12 @@ static struct snd_soc_card mxs_sgtl5000 = { .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; -static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) +static int mxs_sgtl5000_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &mxs_sgtl5000; + int ret, i; struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i; - - if (!np) - return 1; /* no device tree */ saif_np[0] = of_parse_phandle(np, "saif-controllers", 0); saif_np[1] = of_parse_phandle(np, "saif-controllers", 1); @@ -152,18 +150,6 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return 0; -} - -static int mxs_sgtl5000_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &mxs_sgtl5000; - int ret; - - ret = mxs_sgtl5000_probe_dt(pdev); - if (ret < 0) - return ret; - /* * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). * The Sgtl5000 sysclk is derived from saif0 mclk and it's range -- cgit From 89d051300b845e1001a9d9e9ce94da4250c21613 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:22:18 +0530 Subject: ASoC: rt5640: Staticize hp_amp_power_on 'hp_amp_power_on' is used only in this file. Make it static. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index c26a8f814b18..2f6bb161e64c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -926,7 +926,7 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } -void hp_amp_power_on(struct snd_soc_codec *codec) +static void hp_amp_power_on(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); -- cgit From 9e9cb9b99615180b94d743f1d6ca0f82539c8754 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 13 Sep 2013 17:57:35 +0100 Subject: ASoC: rt5640: Provide more useful hw_params error reasons. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 2f6bb161e64c..de40bd9f6ac2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1609,7 +1609,8 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, rt5640->lrck[dai->id] = params_rate(params); pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5640->lrck[dai->id], dai->id); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); -- cgit From 02b80773de3732dae11c1cf0c1ce40378901bd0e Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 13 Sep 2013 17:57:36 +0100 Subject: ASoC: rt5640: Add ACPI probing support. Allow the RT5640 to be probed as an ACPI I2C device. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index de40bd9f6ac2..0bfb960e90f8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -2081,6 +2082,12 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +static struct acpi_device_id rt5640_acpi_match[] = { + { "INT33CA", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); + static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) { rt5640->pdata.in1_diff = of_property_read_bool(np, @@ -2200,6 +2207,7 @@ static struct i2c_driver rt5640_i2c_driver = { .driver = { .name = "rt5640", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5640_acpi_match), }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, -- cgit From 37c83edf9afd3d7b39ace9113a166c03b7a2820f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:17:08 +0100 Subject: ASoC: wm8400: Use supplies to manage input power Rather than using a fake register to manage input power create some supply widgets and use those. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 73 ++++++++++++----------------------------------- 1 file changed, 18 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index d2a092850283..95c33d169952 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -32,13 +32,6 @@ #include "wm8400.h" -/* Fake register for internal state */ -#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1) -#define WM8400_INMIXL_PWR 0 -#define WM8400_AINLMUX_PWR 1 -#define WM8400_INMIXR_PWR 2 -#define WM8400_AINRMUX_PWR 3 - static struct regulator_bulk_data power[] = { { .supply = "I2S1VDD", @@ -79,10 +72,7 @@ static inline unsigned int wm8400_read(struct snd_soc_codec *codec, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) - return wm8400->fake_register; - else - return wm8400_reg_read(wm8400->wm8400, reg); + return wm8400_reg_read(wm8400->wm8400, reg); } /* @@ -93,11 +83,7 @@ static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) { - wm8400->fake_register = value; - return 0; - } else - return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); + return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); } static void wm8400_codec_reset(struct snd_soc_codec *codec) @@ -352,32 +338,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, * _DAPM_ Controls */ -static int inmixer_event (struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); - - if (fakepower & ((1 << WM8400_INMIXL_PWR) | - (1 << WM8400_AINLMUX_PWR))) { - reg |= WM8400_AINL_ENA; - } else { - reg &= ~WM8400_AINL_ENA; - } - - if (fakepower & ((1 << WM8400_INMIXR_PWR) | - (1 << WM8400_AINRMUX_PWR))) { - reg |= WM8400_AINR_ENA; - } else { - reg &= ~WM8400_AINR_ENA; - } - snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); - - return 0; -} - static int outmixer_event (struct snd_soc_dapm_widget *w, struct snd_kcontrol * kcontrol, int event) { @@ -658,27 +618,26 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2, 0, &wm8400_dapm_rin34_pga_controls[0], ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)), +SND_SOC_DAPM_SUPPLY("INL", WM8400_POWER_MANAGEMENT_2, WM8400_AINL_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("INR", WM8400_POWER_MANAGEMENT_2, WM8400_AINR_ENA_SHIFT, + 0, NULL, 0), + /* INMIXL */ -SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixl_controls)), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0, - &wm8400_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AILNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainlmux_controls), /* INMIXR */ -SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixr_controls)), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0, - &wm8400_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AIRNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -789,11 +748,13 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { {"LIN34 PGA", "LIN3 Switch", "LIN3"}, {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ + {"INMIXL", NULL, "INL"}, {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, /* AILNMUX */ + {"AILNMUX", NULL, "INL"}, {"AILNMUX", "INMIXL Mix", "INMIXL"}, {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"}, {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"}, @@ -808,12 +769,14 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, - /* INMIXL */ + /* INMIXR */ + {"INMIXR", NULL, "INR"}, {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, /* AIRNMUX */ + {"AIRNMUX", NULL, "INR"}, {"AIRNMUX", "INMIXR Mix", "INMIXR"}, {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"}, {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"}, -- cgit From b8cc4151f8af97e1b573ca399a77f439f401a57e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:21:12 +0100 Subject: ASoC: wm8400: Use regmap for I/O Since we no longer have a fake register to simulate we can use the framework for I/O. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 95c33d169952..48dc7d2fee36 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -67,25 +67,6 @@ struct wm8400_priv { int fll_in, fll_out; }; -static inline unsigned int wm8400_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - - return wm8400_reg_read(wm8400->wm8400, reg); -} - -/* - * write to the wm8400 register space - */ -static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - - return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); -} - static void wm8400_codec_reset(struct snd_soc_codec *codec) { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); @@ -1328,9 +1309,12 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); - codec->control_data = priv->wm8400 = wm8400; + priv->wm8400 = wm8400; + codec->control_data = wm8400->regmap; priv->codec = codec; + snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = devm_regulator_bulk_get(wm8400->dev, ARRAY_SIZE(power), &power[0]); if (ret != 0) { @@ -1377,8 +1361,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .remove = wm8400_codec_remove, .suspend = wm8400_suspend, .resume = wm8400_resume, - .read = snd_soc_read, - .write = wm8400_write, .set_bias_level = wm8400_set_bias_level, .controls = wm8400_snd_controls, -- cgit From a0ff6ea24f785ec58bccdbce7b366661c57e3591 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 11 Sep 2013 15:27:29 +0100 Subject: ASoC: samsung: Allow mono in i2s driver Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b302f3b7a587..a7e3519ad7c4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1060,7 +1060,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.ops = &samsung_i2s_dai_ops; i2s->i2s_dai_drv.suspend = i2s_suspend; i2s->i2s_dai_drv.resume = i2s_resume; - i2s->i2s_dai_drv.playback.channels_min = 2; + i2s->i2s_dai_drv.playback.channels_min = 1; i2s->i2s_dai_drv.playback.channels_max = 2; i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; -- cgit From b3a6006e1d106fddcfc121d0ccfa9b7faeeb8f3e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Sep 2013 15:38:15 +0100 Subject: ASoC: bells: Add missing route to power up DSP clock Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 29e246803626..84f5d8b76679 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -356,6 +356,7 @@ static struct snd_soc_dapm_widget bells_widgets[] = { static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + { "CLKIN", NULL, "OPCLK" }, { "DMIC", NULL, "MICBIAS2" }, { "IN2L", NULL, "DMIC" }, -- cgit From 49c60547daebaa79e8de9d2dff6dee994576c94c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Sep 2013 15:34:35 +0100 Subject: ASoC: arizona: Improve handling of setting REFCLK to 0 This patch suppresses calculation of REFCLK parameters when the REFCLK source frequency is set to zero, additionally it will consider a source frequency of zero as the REFCLK being disabled and switch to using the SYNCCLK. Reported-by: Kyung Kwee Ryu Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808ba1418..6f05b17d1965 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1477,21 +1477,25 @@ static void arizona_enable_fll(struct arizona_fll *fll, { struct arizona *arizona = fll->arizona; int ret; + bool use_sync = false; /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. */ - if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + if (fll->ref_src >= 0 && fll->ref_freq && + fll->ref_src != fll->sync_src) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); - if (fll->sync_src >= 0) + if (fll->sync_src >= 0) { arizona_apply_fll(arizona, fll->base + 0x10, sync, fll->sync_src, true); + use_sync = true; + } } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, @@ -1511,7 +1515,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, * Increase the bandwidth if we're not using a low frequency * sync source. */ - if (fll->sync_src >= 0 && fll->sync_freq > 100000) + if (use_sync && fll->sync_freq > 100000) regmap_update_bits(arizona->regmap, fll->base + 0x17, ARIZONA_FLL1_SYNC_BW, 0); else @@ -1526,8 +1530,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->ref_src >= 0 && fll->sync_src >= 0 && - fll->ref_src != fll->sync_src) + if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1561,10 +1564,12 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout && Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; + if (fll->fout) { + if (Fref > 0) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); + if (ret != 0) + return ret; + } if (fll->sync_src >= 0) { ret = arizona_calc_fll(fll, &sync, fll->sync_freq, -- cgit From 193b2f65b87e9da78b15f3e3a0cae1d37fbafa57 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 16 Sep 2013 18:14:20 +0200 Subject: ASoC: ak4104: provide a module device table Provide a module device table for the SPI subsystem, so the driver can be autoloaded by the SPI core. While at it, get rid of an unnecessary #define. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 71059c07ae7b..b4819dcd4f4d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -45,8 +45,6 @@ #define AK4104_TX_TXE (1 << 0) #define AK4104_TX_V (1 << 1) -#define DRV_NAME "ak4104-codec" - struct ak4104_private { struct regmap *regmap; }; @@ -291,12 +289,19 @@ static const struct of_device_id ak4104_of_match[] = { }; MODULE_DEVICE_TABLE(of, ak4104_of_match); +static const struct spi_device_id ak4104_id_table[] = { + { "ak4104", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, ak4104_id_table); + static struct spi_driver ak4104_spi_driver = { .driver = { - .name = DRV_NAME, + .name = "ak4104", .owner = THIS_MODULE, .of_match_table = ak4104_of_match, }, + .id_table = ak4104_id_table, .probe = ak4104_spi_probe, .remove = ak4104_spi_remove, }; -- cgit From bf551413038f74343ec4d1413c3610e2362d0aeb Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 16:16:17 +0530 Subject: ASoC: twl6040: Remove redundant semicolon Redundant semicolon removed. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb6c323..35059a242fa4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) -- cgit From a0b03a616b08cf9d709812ff5cf7e9c0958d6807 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Sep 2013 20:37:34 +0100 Subject: ASoC: core: Implement devm_snd_soc_register_component() Since with the wider use of devres many drivers are now only calling snd_soc_unregister_component() in their remove functions providing a managed version will save a reasonable amount of code. Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/Makefile | 2 +- sound/soc/soc-devres.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 56 insertions(+), 1 deletion(-) create mode 100644 sound/soc/soc-devres.c diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..b970f019b452 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -386,6 +386,9 @@ void snd_soc_unregister_codec(struct device *dev); int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, unsigned int reg); diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 61a64d281905..8b9e70105dd2 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c new file mode 100644 index 000000000000..13fe86f7c9a8 --- /dev/null +++ b/sound/soc/soc-devres.c @@ -0,0 +1,52 @@ +/* + * soc-devres.c -- ALSA SoC Audio Layer devres functions + * + * Copyright (C) 2013 Linaro Ltd + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include + +static void devm_component_release(struct device *dev, void *res) +{ + snd_soc_unregister_component(*(struct device **)res); +} + +/** + * devm_snd_soc_register_component - resource managed component registration + * @dev: Device used to manage component + * @cmpnt_drv: Component driver + * @dai_drv: DAI driver + * @num_dai: Number of DAIs to register + * + * Register a component with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); -- cgit From 0e4ff5c806263bf40ee5409ac283b776f0c11e41 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Sep 2013 18:02:05 +0100 Subject: ASoC: core: Add devm_snd_soc_register_card() Simplify error handling and remove repetitive (and rarely executed) code for unregistration by providing a devm_snd_soc_register() card. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 1 + sound/soc/soc-devres.c | 34 ++++++++++++++++++++++++++++++++++ 2 files changed, 35 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index b970f019b452..d44728ab2be0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -369,6 +369,7 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 13fe86f7c9a8..b1d732255c02 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -50,3 +50,37 @@ int devm_snd_soc_register_component(struct device *dev, return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); + +static void devm_card_release(struct device *dev, void *res) +{ + snd_soc_unregister_card(*(struct snd_soc_card **)res); +} + +/** + * devm_snd_soc_register_card - resource managed card registration + * @dev: Device used to manage card + * @card: Card to register + * + * Register a card with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_card(card); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); -- cgit From 9a8e0322f0a8c7506f4ced07ec2a7e7e2a9cbe4a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Sep 2013 18:02:28 +0100 Subject: ASoC: smdk_wm8994: Use devm_snd_soc_unregister_card() Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 5fd7a05a9b9e..831972d24fb9 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -193,7 +193,7 @@ static int smdk_audio_probe(struct platform_device *pdev) platform_set_drvdata(pdev, board); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); @@ -201,15 +201,6 @@ static int smdk_audio_probe(struct platform_device *pdev) return ret; } -static int smdk_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver smdk_audio_driver = { .driver = { .name = "smdk-audio-wm8894", @@ -217,7 +208,6 @@ static struct platform_driver smdk_audio_driver = { .of_match_table = of_match_ptr(samsung_wm8994_of_match), }, .probe = smdk_audio_probe, - .remove = smdk_audio_remove, }; module_platform_driver(smdk_audio_driver); -- cgit From d644a115e86433abbb544808c4be1e4b5a048c2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Sep 2013 20:37:51 +0100 Subject: ASoC: samsung-i2s: Use devm_snd_soc_register_component() Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a7e3519ad7c4..32956df8f50c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1143,9 +1143,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Unable to get drvdata\n"); return -EFAULT; } - snd_soc_register_component(&sec_dai->pdev->dev, - &samsung_i2s_component, - &sec_dai->i2s_dai_drv, 1); + devm_snd_soc_register_component(&sec_dai->pdev->dev, + &samsung_i2s_component, + &sec_dai->i2s_dai_drv, 1); samsung_asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1258,8 +1258,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) goto err; } - snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component, - &pri_dai->i2s_dai_drv, 1); + devm_snd_soc_register_component(&pri_dai->pdev->dev, + &samsung_i2s_component, + &pri_dai->i2s_dai_drv, 1); pm_runtime_enable(&pdev->dev); @@ -1294,7 +1295,6 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->sec_dai = NULL; samsung_asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- cgit From 0feb23d1bdf31db903069d3d94892e56b5c11981 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:50:50 +0530 Subject: ASoC: ak4642: Remove redundant break 'break' after return statement is redundant. Remove it. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d0378709702..21c35ed778cc 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -352,7 +352,6 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) */ default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); @@ -405,7 +404,6 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, break; default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); -- cgit From e54cf76ba2c9ec071a68e98f2830226c0cac8086 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 16 Sep 2013 13:01:46 +0100 Subject: ASoC: core: Add API for configuration of DAI BCLK ratio Some codec drivers when running in slave mode require that BCLK to sample rate ratio is explicitly set by the machine driver as it may not be exactly rate * frame size. Extend the DAI API by adding :- int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 +++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227d35d3..d8acf0ca77be 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -105,6 +105,8 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); + /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); @@ -131,6 +133,7 @@ struct snd_soc_dai_ops { int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); /* * DAI format configuration diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d0561312f3b..31adad04222d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3576,6 +3576,22 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); +/** + * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. + * @dai: DAI + * @ratio Ratio of BCLK to Sample rate. + * + * Configures the DAI for a preset BCLK to sample rate ratio. + */ +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + if (dai->driver && dai->driver->ops->set_bclk_ratio) + return dai->driver->ops->set_bclk_ratio(dai, ratio); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); + /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI -- cgit From d191bd8de8c61619563f2b19f1fdcc0944ff1a72 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 4 Sep 2013 19:39:03 -0700 Subject: ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 33 +++++---- sound/soc/soc-core.c | 202 ++++++++++++++++++++++++++++----------------------- 2 files changed, 131 insertions(+), 104 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..9a81e2e7d661 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -670,6 +670,21 @@ struct snd_soc_cache_ops { int (*sync)(struct snd_soc_codec *codec); }; +/* component interface */ +struct snd_soc_component_driver { + const char *name; +}; + +struct snd_soc_component { + const char *name; + int id; + int num_dai; + struct device *dev; + struct list_head list; + + const struct snd_soc_component_driver *driver; +}; + /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; @@ -715,6 +730,9 @@ struct snd_soc_codec { struct mutex cache_rw_mutex; int val_bytes; + /* component */ + struct snd_soc_component component; + /* dapm */ struct snd_soc_dapm_context dapm; unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ @@ -733,6 +751,7 @@ struct snd_soc_codec_driver { int (*remove)(struct snd_soc_codec *); int (*suspend)(struct snd_soc_codec *); int (*resume)(struct snd_soc_codec *); + struct snd_soc_component_driver component_driver; /* Default control and setup, added after probe() is run */ const struct snd_kcontrol_new *controls; @@ -849,20 +868,6 @@ struct snd_soc_platform { #endif }; -struct snd_soc_component_driver { - const char *name; -}; - -struct snd_soc_component { - const char *name; - int id; - int num_dai; - struct device *dev; - struct list_head list; - - const struct snd_soc_component_driver *driver; -}; - struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d0561312f3b..014ac10267da 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4020,6 +4020,112 @@ static void snd_soc_unregister_dais(struct device *dev, size_t count) snd_soc_unregister_dai(dev); } +/** + * snd_soc_register_component - Register a component with the ASoC core + * + */ +static int +__snd_soc_register_component(struct device *dev, + struct snd_soc_component *cmpnt, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai, bool allow_single_dai) +{ + int ret; + + dev_dbg(dev, "component register %s\n", dev_name(dev)); + + if (!cmpnt) { + dev_err(dev, "ASoC: Failed to connecting component\n"); + return -ENOMEM; + } + + cmpnt->name = fmt_single_name(dev, &cmpnt->id); + if (!cmpnt->name) { + dev_err(dev, "ASoC: Failed to simplifying name\n"); + return -ENOMEM; + } + + cmpnt->dev = dev; + cmpnt->driver = cmpnt_drv; + cmpnt->num_dai = num_dai; + + /* + * snd_soc_register_dai() uses fmt_single_name(), and + * snd_soc_register_dais() uses fmt_multiple_name() + * for dai->name which is used for name based matching + * + * this function is used from cpu/codec. + * allow_single_dai flag can ignore "codec" driver reworking + * since it had been used snd_soc_register_dais(), + */ + if ((1 == num_dai) && allow_single_dai) + ret = snd_soc_register_dai(dev, dai_drv); + else + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto error_component_name; + } + + mutex_lock(&client_mutex); + list_add(&cmpnt->list, &component_list); + mutex_unlock(&client_mutex); + + dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name); + + return ret; + +error_component_name: + kfree(cmpnt->name); + + return ret; +} + +int snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai) +{ + struct snd_soc_component *cmpnt; + + cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL); + if (!cmpnt) { + dev_err(dev, "ASoC: Failed to allocate memory\n"); + return -ENOMEM; + } + + return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, + dai_drv, num_dai, true); +} +EXPORT_SYMBOL_GPL(snd_soc_register_component); + +/** + * snd_soc_unregister_component - Unregister a component from the ASoC core + * + */ +void snd_soc_unregister_component(struct device *dev) +{ + struct snd_soc_component *cmpnt; + + list_for_each_entry(cmpnt, &component_list, list) { + if (dev == cmpnt->dev) + goto found; + } + return; + +found: + snd_soc_unregister_dais(dev, cmpnt->num_dai); + + mutex_lock(&client_mutex); + list_del(&cmpnt->list); + mutex_unlock(&client_mutex); + + dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); + kfree(cmpnt->name); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_component); + /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -4242,10 +4348,12 @@ int snd_soc_register_codec(struct device *dev, list_add(&codec->list, &codec_list); mutex_unlock(&client_mutex); - /* register any DAIs */ - ret = snd_soc_register_dais(dev, dai_drv, num_dai); + /* register component */ + ret = __snd_soc_register_component(dev, &codec->component, + &codec_drv->component_driver, + dai_drv, num_dai, false); if (ret < 0) { - dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret); goto fail_codec_name; } @@ -4280,7 +4388,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - snd_soc_unregister_dais(dev, codec->num_dai); + snd_soc_unregister_component(dev); mutex_lock(&client_mutex); list_del(&codec->list); @@ -4295,92 +4403,6 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); - -/** - * snd_soc_register_component - Register a component with the ASoC core - * - */ -int snd_soc_register_component(struct device *dev, - const struct snd_soc_component_driver *cmpnt_drv, - struct snd_soc_dai_driver *dai_drv, - int num_dai) -{ - struct snd_soc_component *cmpnt; - int ret; - - dev_dbg(dev, "component register %s\n", dev_name(dev)); - - cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL); - if (!cmpnt) { - dev_err(dev, "ASoC: Failed to allocate memory\n"); - return -ENOMEM; - } - - cmpnt->name = fmt_single_name(dev, &cmpnt->id); - if (!cmpnt->name) { - dev_err(dev, "ASoC: Failed to simplifying name\n"); - return -ENOMEM; - } - - cmpnt->dev = dev; - cmpnt->driver = cmpnt_drv; - cmpnt->num_dai = num_dai; - - /* - * snd_soc_register_dai() uses fmt_single_name(), and - * snd_soc_register_dais() uses fmt_multiple_name() - * for dai->name which is used for name based matching - */ - if (1 == num_dai) - ret = snd_soc_register_dai(dev, dai_drv); - else - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); - goto error_component_name; - } - - mutex_lock(&client_mutex); - list_add(&cmpnt->list, &component_list); - mutex_unlock(&client_mutex); - - dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name); - - return ret; - -error_component_name: - kfree(cmpnt->name); - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_register_component); - -/** - * snd_soc_unregister_component - Unregister a component from the ASoC core - * - */ -void snd_soc_unregister_component(struct device *dev) -{ - struct snd_soc_component *cmpnt; - - list_for_each_entry(cmpnt, &component_list, list) { - if (dev == cmpnt->dev) - goto found; - } - return; - -found: - snd_soc_unregister_dais(dev, cmpnt->num_dai); - - mutex_lock(&client_mutex); - list_del(&cmpnt->list); - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); - kfree(cmpnt->name); -} -EXPORT_SYMBOL_GPL(snd_soc_unregister_component); - /* Retrieve a card's name from device tree */ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname) -- cgit From cb470087669a3fab1958fec79dd7db280b33f178 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Sep 2013 17:39:56 -0700 Subject: ASoC: add .of_xlate_dai_name on snd_soc_component_driver ASoC sound driver requires CPU/CODEC drivers for probing, and each CPU/CODEC has some DAI on it. Then, "dai name matching" have been used to identify CPU-CODEC DAI pair on ASoC. But, the "dai port number matching" is now required from DeviceTree. The solution of this issue is to replace the dai port number into dai name. Now, CPU/CODEC are based on struct snd_soc_component, and it can care above as common issue. This patch adds .of_xlate_dai_name callback interface on struct snd_soc_component_driver, and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name. Then, #sound-dai-cells which enables DAI specifier is required on CPU/CODEC device tree properties. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 35 +++++++++++++++++++++++++++++++++++ 2 files changed, 43 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a81e2e7d661..1dd7dc5f7d52 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -13,6 +13,7 @@ #ifndef __LINUX_SND_SOC_H #define __LINUX_SND_SOC_H +#include #include #include #include @@ -673,6 +674,11 @@ struct snd_soc_cache_ops { /* component interface */ struct snd_soc_component_driver { const char *name; + + /* DT */ + int (*of_xlate_dai_name)(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name); }; struct snd_soc_component { @@ -1206,6 +1212,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); +int snd_soc_of_get_dai_name(struct device_node *of_node, + const char **dai_name); #include diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 014ac10267da..711bd362028d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4590,6 +4590,41 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); +int snd_soc_of_get_dai_name(struct device_node *of_node, + const char **dai_name) +{ + struct snd_soc_component *pos; + struct of_phandle_args args; + int ret; + + ret = of_parse_phandle_with_args(of_node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) + return ret; + + ret = -EPROBE_DEFER; + + mutex_lock(&client_mutex); + list_for_each_entry(pos, &component_list, list) { + if (pos->dev->of_node != args.np) + continue; + + if (!pos->driver->of_xlate_dai_name) { + ret = -ENOSYS; + break; + } + + ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name); + break; + } + mutex_unlock(&client_mutex); + + of_node_put(args.np); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- cgit From e19bcb6b95c0326ca364814b86b32799aa7e20db Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:52:42 +0530 Subject: ASoC: fsl_spdif: Remove redundant semicolon Redundant semicolon at the end of brace is removed. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e849ce..c0fea02114e1 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { -- cgit From feb8f1147618ebf20ab3e5efc143ceb621063f81 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:59:25 +0530 Subject: ASoC: fsl_ssi: Remove redundant dev_set_drvdata Driver core sets the driver data to NULL on detach. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4973be774956..6ac87300d45d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1135,7 +1135,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); -- cgit From 072188b61c9b7aedaa15c46226b537345644beee Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 1 Sep 2013 20:31:16 -0700 Subject: ASoC: rsnd: gen: rsnd_gen_ops cares .probe and .remove Current rsnd_gen_ops didn't care about .probe and .remove functions, but it was not good sense. This patch tidyup it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 41 ++++++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index babb203b43b7..331fc558d796 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -11,6 +11,11 @@ #include "rsnd.h" struct rsnd_gen_ops { + int (*probe)(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); + void (*remove)(struct platform_device *pdev, + struct rsnd_priv *priv); int (*path_init)(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io); @@ -98,11 +103,6 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv, return ret; } -static struct rsnd_gen_ops rsnd_gen1_ops = { - .path_init = rsnd_gen1_path_init, - .path_exit = rsnd_gen1_path_exit, -}; - #define RSND_GEN1_REG_MAP(g, s, i, oi, oa) \ do { \ (g)->reg_map[RSND_REG_##i].index = RSND_GEN1_##s; \ @@ -163,7 +163,6 @@ static int rsnd_gen1_probe(struct platform_device *pdev, IS_ERR(gen->base[RSND_GEN1_SSI])) return -ENODEV; - gen->ops = &rsnd_gen1_ops; rsnd_gen1_reg_map_init(gen); dev_dbg(dev, "Gen1 device probed\n"); @@ -183,6 +182,13 @@ static void rsnd_gen1_remove(struct platform_device *pdev, { } +static struct rsnd_gen_ops rsnd_gen1_ops = { + .probe = rsnd_gen1_probe, + .remove = rsnd_gen1_remove, + .path_init = rsnd_gen1_path_init, + .path_exit = rsnd_gen1_path_exit, +}; + /* * Gen */ @@ -251,6 +257,14 @@ int rsnd_gen_probe(struct platform_device *pdev, return -ENOMEM; } + if (rsnd_is_gen1(priv)) + gen->ops = &rsnd_gen1_ops; + + if (!gen->ops) { + dev_err(dev, "unknown generation R-Car sound device\n"); + return -ENODEV; + } + priv->gen = gen; /* @@ -261,20 +275,13 @@ int rsnd_gen_probe(struct platform_device *pdev, for (i = 0; i < RSND_REG_MAX; i++) gen->reg_map[i].index = -1; - /* - * init each module - */ - if (rsnd_is_gen1(priv)) - return rsnd_gen1_probe(pdev, info, priv); - - dev_err(dev, "unknown generation R-Car sound device\n"); - - return -ENODEV; + return gen->ops->probe(pdev, info, priv); } void rsnd_gen_remove(struct platform_device *pdev, struct rsnd_priv *priv) { - if (rsnd_is_gen1(priv)) - rsnd_gen1_remove(pdev, priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + gen->ops->remove(pdev, priv); } -- cgit From bcf25567ecec6cb0a8078cbf68969baed047fdf4 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 17 Sep 2013 12:26:00 +0300 Subject: ASoC: davinci-evm: Move sysclk logic away from evm_hw_params The sysclk rate does not change runtime so it should be initialized at init time. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 64 ++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 20 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fd7c45b9ed5a..2f8161c1d5f0 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -27,6 +27,10 @@ #include "davinci-i2s.h" #include "davinci-mcasp.h" +struct snd_soc_card_drvdata_davinci { + unsigned sysclk; +}; + #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) static int evm_hw_params(struct snd_pcm_substream *substream, @@ -35,27 +39,11 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *soc_card = codec->card; int ret = 0; - unsigned sysclk; - - /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || - machine_is_davinci_dm365_evm()) - sysclk = 27000000; - - /* ASP0 in DM6446 EVM is clocked by U55, as configured by - * board-dm644x-evm.c using GPIOs from U18. There are six - * options; here we "know" we use a 48 KHz sample rate. - */ - else if (machine_is_davinci_evm()) - sysclk = 12288000; - - else if (machine_is_davinci_da830_evm() || - machine_is_davinci_da850_evm()) - sysclk = 24576000; - - else - return -EINVAL; + unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) + snd_soc_card_get_drvdata(soc_card))->sysclk; /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); @@ -243,35 +231,65 @@ static struct snd_soc_dai_link da850_evm_dai = { }; /* davinci dm6446 evm audio machine driver */ +/* + * ASP0 in DM6446 EVM is clocked by U55, as configured by + * board-dm644x-evm.c using GPIOs from U18. There are six + * options; here we "know" we use a 48 KHz sample rate. + */ +static struct snd_soc_card_drvdata_davinci dm6446_snd_soc_card_drvdata = { + .sysclk = 12288000, +}; + static struct snd_soc_card dm6446_snd_soc_card_evm = { .name = "DaVinci DM6446 EVM", .owner = THIS_MODULE, .dai_link = &dm6446_evm_dai, .num_links = 1, + .drvdata = &dm6446_snd_soc_card_drvdata, }; /* davinci dm355 evm audio machine driver */ +/* ASP1 on DM355 EVM is clocked by an external oscillator */ +static struct snd_soc_card_drvdata_davinci dm355_snd_soc_card_drvdata = { + .sysclk = 27000000, +}; + static struct snd_soc_card dm355_snd_soc_card_evm = { .name = "DaVinci DM355 EVM", .owner = THIS_MODULE, .dai_link = &dm355_evm_dai, .num_links = 1, + .drvdata = &dm355_snd_soc_card_drvdata, }; /* davinci dm365 evm audio machine driver */ +static struct snd_soc_card_drvdata_davinci dm365_snd_soc_card_drvdata = { + .sysclk = 27000000, +}; + static struct snd_soc_card dm365_snd_soc_card_evm = { .name = "DaVinci DM365 EVM", .owner = THIS_MODULE, .dai_link = &dm365_evm_dai, .num_links = 1, + .drvdata = &dm365_snd_soc_card_drvdata, }; /* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card_drvdata_davinci dm6467_snd_soc_card_drvdata = { + .sysclk = 27000000, +}; + static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", .owner = THIS_MODULE, .dai_link = dm6467_evm_dai, .num_links = ARRAY_SIZE(dm6467_evm_dai), + .drvdata = &dm6467_snd_soc_card_drvdata, +}; + +static struct snd_soc_card_drvdata_davinci da830_snd_soc_card_drvdata = { + .sysclk = 24576000, }; static struct snd_soc_card da830_snd_soc_card = { @@ -279,6 +297,11 @@ static struct snd_soc_card da830_snd_soc_card = { .owner = THIS_MODULE, .dai_link = &da830_evm_dai, .num_links = 1, + .drvdata = &da830_snd_soc_card_drvdata, +}; + +static struct snd_soc_card_drvdata_davinci da850_snd_soc_card_drvdata = { + .sysclk = 24576000, }; static struct snd_soc_card da850_snd_soc_card = { @@ -286,6 +309,7 @@ static struct snd_soc_card da850_snd_soc_card = { .owner = THIS_MODULE, .dai_link = &da850_evm_dai, .num_links = 1, + .drvdata = &da850_snd_soc_card_drvdata, }; static struct platform_device *evm_snd_device; -- cgit From 5fb7680bd0035525eb1534001f7b7f2ca06a8ab7 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 10:41:36 +0530 Subject: ASoC: SPEAr spdif_in: Use devm_snd_soc_register_component devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat Acked-by: Viresh Kumar Signed-off-by: Mark Brown --- sound/soc/spear/spdif_in.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 63acfeb4b69d..21a8c954af1c 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -257,20 +257,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); -} - -static int spdif_in_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); } static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, - .remove = spdif_in_remove, .driver = { .name = "spdif-in", .owner = THIS_MODULE, -- cgit From 77aea716872fd976cbb9706b4588cf1fb9d52826 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 10:41:37 +0530 Subject: ASoC: SPEAr spdif_out: Use devm_snd_soc_register_component devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat Acked-by: Viresh Kumar Signed-off-by: Mark Brown --- sound/soc/spear/spdif_out.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 2fdf68c98d22..70fc4d687529 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -307,18 +307,11 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); return ret; } -static int spdif_out_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; -} - #ifdef CONFIG_PM static int spdif_out_suspend(struct device *dev) { @@ -357,7 +350,6 @@ static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \ static struct platform_driver spdif_out_driver = { .probe = spdif_out_probe, - .remove = spdif_out_remove, .driver = { .name = "spdif-out", .owner = THIS_MODULE, -- cgit From be7c5926c22403b6f5895a10a73145925dd560a9 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 17 Sep 2013 12:26:06 +0300 Subject: ASoC: tlv320aic3x: Add regulators to DT bindings document Add regulator properties to tlv320aic3x DT bindings document. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 705a6b156c6c..ba2647751aa5 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -24,10 +24,17 @@ Optional properties: 3 - MICBIAS output is connected to AVDD, If this node is not mentioned or if the value is incorrect, then MicBias is powered down. +- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the + device as covered in Documentation/devicetree/bindings/regulator/regulator.txt Example: tlv320aic3x: tlv320aic3x@1b { compatible = "ti,tlv320aic3x"; reg = <0x1b>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; }; -- cgit From c9f3de41b93ae61da4cd10611e446a4b916245d2 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 17 Sep 2013 12:26:07 +0300 Subject: ASoC: tlv320aic3x: Add codec pins to DT bindings document Add list of codec pins to tlv320aic3x DT bindings document. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tlv320aic3x.txt | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index ba2647751aa5..5e6040c2c2e9 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -27,6 +27,25 @@ Optional properties: - AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the device as covered in Documentation/devicetree/bindings/regulator/regulator.txt +CODEC output pins: + * LLOUT + * RLOUT + * MONO_LOUT + * HPLOUT + * HPROUT + * HPLCOM + * HPRCOM + +CODEC input pins: + * MIC3L + * MIC3R + * LINE1L + * LINE2L + * LINE1R + * LINE2R + +The pins can be used in referring sound node's audio-routing property. + Example: tlv320aic3x: tlv320aic3x@1b { -- cgit From 32fcb97b9f699f63742bcaadca6e0beede86e8e8 Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Thu, 19 Sep 2013 11:18:06 +0200 Subject: ASoC: rt5640: Omit ACPI match table only if !ACPI The ACPI_PTR() macro evaluates to NULL if ACPI is disabled and hence the ACPI match table won't be used, causing the compiler to complain. Avoid this by protecting the table using an #ifdef CONFIG_ACPI. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0bfb960e90f8..641eeeb00c5c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2082,11 +2082,13 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); +#endif static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) { -- cgit From 25db0dc88016ec67ec4e38164482a3d7b7429f75 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 19 Sep 2013 12:23:44 +0530 Subject: ASoC: SPEAr spdif_out: Remove redundant variable Return directly and remove the intermediate local variable. Signed-off-by: Sachin Kamat Acked-by: Viresh Kumar Signed-off-by: Mark Brown --- sound/soc/spear/spdif_out.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 70fc4d687529..b6ef6f78dc78 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -280,7 +280,6 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; - int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -307,9 +306,8 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); - return ret; + return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); } #ifdef CONFIG_PM -- cgit From 01984a47e21a7d36cea0d6c0933c8173391721fc Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 09:42:46 +0530 Subject: ASoC: imx-sgtl5000: Use devm_snd_soc_register_card devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4fdfc52..78f86d870b11 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -159,7 +159,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; @@ -180,15 +180,6 @@ fail: return ret; } -static int imx_sgtl5000_remove(struct platform_device *pdev) -{ - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - - snd_soc_unregister_card(&data->card); - - return 0; -} - static const struct of_device_id imx_sgtl5000_dt_ids[] = { { .compatible = "fsl,imx-audio-sgtl5000", }, { /* sentinel */ } @@ -202,7 +193,6 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, - .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); -- cgit From ff27d9b3d6dd26013537e4f8162627169ca92af4 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 09:42:47 +0530 Subject: ASoC: imx-spdif: Use devm_snd_soc_register_card devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/fsl/imx-spdif.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 816013b0ebba..8499d5292f08 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -87,7 +87,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) if (ret) goto error_dir; - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); goto error_dir; @@ -119,8 +119,6 @@ static int imx_spdif_audio_remove(struct platform_device *pdev) if (data->txdev) platform_device_unregister(data->txdev); - snd_soc_unregister_card(&data->card); - return 0; } -- cgit From eafbae2919e24b6ed7078464b2c8b2313ac0252c Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 09:42:48 +0530 Subject: ASoC: imx-wm8962: Use devm_snd_soc_register_card devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/fsl/imx-wm8962.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 722afe69169e..6c6066618f3b 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -266,7 +266,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; @@ -296,7 +296,6 @@ static int imx_wm8962_remove(struct platform_device *pdev) if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); - snd_soc_unregister_card(&data->card); return 0; } -- cgit From 33c89c30afd7b6a309638d8d9b88481118d0a3ec Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 09:49:04 +0530 Subject: ASoC: mfld: Use devm_snd_soc_register_card devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index ee363845759e..d3d4c32434f7 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -400,7 +400,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; - ret_val = snd_soc_register_card(&snd_soc_card_mfld); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); if (ret_val) { pr_debug("snd_soc_register_card failed %d\n", ret_val); return ret_val; @@ -410,20 +410,12 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) return 0; } -static int snd_mfld_mc_remove(struct platform_device *pdev) -{ - pr_debug("snd_mfld_mc_remove called\n"); - snd_soc_unregister_card(&snd_soc_card_mfld); - return 0; -} - static struct platform_driver snd_mfld_mc_driver = { .driver = { .owner = THIS_MODULE, .name = "msic_audio", }, .probe = snd_mfld_mc_probe, - .remove = snd_mfld_mc_remove, }; module_platform_driver(snd_mfld_mc_driver); -- cgit From 67a48b8181b0c981c59acf06f056e00184d3debd Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 09:53:59 +0530 Subject: ASoC: omap-twl4030: Use devm_snd_soc_register_card devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/omap/omap-twl4030.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 2a9324f794d8..6a8d6b5f160d 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -338,9 +338,9 @@ static int omap_twl4030_probe(struct platform_device *pdev) } snd_soc_card_set_drvdata(card, priv); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", ret); return ret; } @@ -357,7 +357,6 @@ static int omap_twl4030_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&priv->hs_jack, ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios); - snd_soc_unregister_card(card); return 0; } -- cgit From 256218ae65d2e59ef5d257355791a62af7d31b3c Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 10:13:49 +0530 Subject: ASoC: fsl_spdif: Use devm_snd_soc_register_component devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e849ce..44378e6e2696 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1172,23 +1172,16 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Register with ASoC */ dev_set_drvdata(&pdev->dev, spdif_priv); - ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component, - &spdif_priv->cpu_dai_drv, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); return ret; } ret = imx_pcm_dma_init(pdev); - if (ret) { + if (ret) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); - goto error_component; - } - - return ret; - -error_component: - snd_soc_unregister_component(&pdev->dev); return ret; } @@ -1196,7 +1189,6 @@ error_component: static int fsl_spdif_remove(struct platform_device *pdev) { imx_pcm_dma_exit(pdev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- cgit From fcd70eb50e2d572a67839410aa30f6b545355980 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 10:20:09 +0530 Subject: ASoC: mxs-saif: Use devm_snd_soc_register_component devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b56b8a0e8deb..14152f6f70dd 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -768,8 +768,8 @@ static int mxs_saif_probe(struct platform_device *pdev) dev_warn(&pdev->dev, "failed to init clocks\n"); } - ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, - &mxs_saif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &mxs_saif_component, + &mxs_saif_dai, 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); return ret; @@ -778,21 +778,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = mxs_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto failed_pdev_alloc; + return ret; } return 0; - -failed_pdev_alloc: - snd_soc_unregister_component(&pdev->dev); - - return ret; } static int mxs_saif_remove(struct platform_device *pdev) { mxs_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- cgit From 6c3cc302a48acdd3797d694ae4a2be82bb71a05a Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 10:28:02 +0530 Subject: ASoC: omap-mcpdm: Use devm_snd_soc_register_component devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 90d2a7cd2563..cd9ee167959d 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -490,14 +490,9 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - return snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, - &omap_mcpdm_dai, 1); -} - -static int asoc_mcpdm_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &omap_mcpdm_component, + &omap_mcpdm_dai, 1); } static const struct of_device_id omap_mcpdm_of_match[] = { @@ -514,7 +509,6 @@ static struct platform_driver asoc_mcpdm_driver = { }, .probe = asoc_mcpdm_probe, - .remove = asoc_mcpdm_remove, }; module_platform_driver(asoc_mcpdm_driver); -- cgit From 9ff50721e47ab0abb8b93159170f67262886ef0d Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 17 Sep 2013 10:32:48 +0530 Subject: ASoC: mmp-sspa: Use devm_snd_soc_register_component devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-sspa.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 41752a5fe3b0..5bf5f1f7cac5 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -455,8 +455,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) priv->dai_fmt = (unsigned int) -1; platform_set_drvdata(pdev, priv); - return snd_soc_register_component(&pdev->dev, &mmp_sspa_component, - &mmp_sspa_dai, 1); + return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component, + &mmp_sspa_dai, 1); } static int asoc_mmp_sspa_remove(struct platform_device *pdev) @@ -466,7 +466,6 @@ static int asoc_mmp_sspa_remove(struct platform_device *pdev) clk_disable(priv->audio_clk); clk_put(priv->audio_clk); clk_put(priv->sysclk); - snd_soc_unregister_component(&pdev->dev); return 0; } -- cgit From fa129ebeba6db2b4bcea45efe87a71d68181c04c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 18:20:26 +0100 Subject: ASoC: 88pm60x: Don't use control data for i2c In preparation for using the regmap directly in the CODEC driver replace references to the I2C client using control_data with references to the driver private data. Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 8af04343cc1a..3925cf34f751 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1166,6 +1166,7 @@ static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, static int pm860x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); int data; switch (level) { @@ -1179,17 +1180,17 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); } break; case SND_SOC_BIAS_OFF: data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_set_bits(codec->control_data, REG_MISC2, data, 0); + pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0); break; } codec->dapm.bias_level = level; @@ -1319,17 +1320,17 @@ int pm860x_hs_jack_detect(struct snd_soc_codec *codec, pm860x->det.lo_shrt = lo_shrt; if (det & SND_JACK_HEADPHONE) - pm860x_set_bits(codec->control_data, REG_HS_DET, + pm860x_set_bits(pm860x->i2c, REG_HS_DET, EN_HS_DET, EN_HS_DET); /* headset short detect */ if (hs_shrt) { data = CLR_SHORT_HS2 | CLR_SHORT_HS1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* Lineout short detect */ if (lo_shrt) { data = CLR_SHORT_LO2 | CLR_SHORT_LO1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* sync status */ @@ -1347,7 +1348,7 @@ int pm860x_mic_jack_detect(struct snd_soc_codec *codec, pm860x->det.mic_det = det; if (det & SND_JACK_MICROPHONE) - pm860x_set_bits(codec->control_data, REG_MIC_DET, + pm860x_set_bits(pm860x->i2c, REG_MIC_DET, MICDET_MASK, MICDET_MASK); /* sync status */ @@ -1377,7 +1378,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE, + ret = pm860x_bulk_read(pm860x->i2c, REG_CACHE_BASE, REG_CACHE_SIZE, codec->reg_cache); if (ret < 0) { dev_err(codec->dev, "Failed to fill register cache: %d\n", -- cgit From f9ded3b2e761256301ebb8d90e87eb1b5443e3ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 19:00:46 +0100 Subject: ASoC: 88pm860x: Use regmap for I/O As part of a move to remove the duplication of regmap functionality in ASoC convert the 88pm860x driver to use the regmap from the MFD. This means that we no longer cache the registers so performance will be slightly reduced on I/O operations. Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 60 ++----------------- sound/soc/codecs/88pm860x-codec.h | 117 +++++++++++++++++++------------------- 2 files changed, 63 insertions(+), 114 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 3925cf34f751..4633e51b1500 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -140,6 +141,7 @@ struct pm860x_priv { unsigned int filter; struct snd_soc_codec *codec; struct i2c_client *i2c; + struct regmap *regmap; struct pm860x_chip *chip; struct pm860x_det det; @@ -269,48 +271,6 @@ static struct st_gain st_table[] = { { -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0}, }; -static int pm860x_volatile(unsigned int reg) -{ - BUG_ON(reg >= REG_CACHE_SIZE); - - switch (reg) { - case PM860X_AUDIO_SUPPLIES_2: - return 1; - } - - return 0; -} - -static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - unsigned char *cache = codec->reg_cache; - - BUG_ON(reg >= REG_CACHE_SIZE); - - if (pm860x_volatile(reg)) - return cache[reg]; - - reg += REG_CACHE_BASE; - - return pm860x_reg_read(codec->control_data, reg); -} - -static int pm860x_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - unsigned char *cache = codec->reg_cache; - - BUG_ON(reg >= REG_CACHE_SIZE); - - if (!pm860x_volatile(reg)) - cache[reg] = (unsigned char)value; - - reg += REG_CACHE_BASE; - - return pm860x_reg_write(codec->control_data, reg, value); -} - static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1364,7 +1324,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; - codec->control_data = pm860x->i2c; + codec->control_data = pm860x->regmap; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, @@ -1378,14 +1338,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = pm860x_bulk_read(pm860x->i2c, REG_CACHE_BASE, - REG_CACHE_SIZE, codec->reg_cache); - if (ret < 0) { - dev_err(codec->dev, "Failed to fill register cache: %d\n", - ret); - goto out; - } - return 0; out: @@ -1408,10 +1360,6 @@ static int pm860x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pm860x = { .probe = pm860x_probe, .remove = pm860x_remove, - .read = pm860x_read_reg_cache, - .write = pm860x_write_reg_cache, - .reg_cache_size = REG_CACHE_SIZE, - .reg_word_size = sizeof(u8), .set_bias_level = pm860x_set_bias_level, .controls = pm860x_snd_controls, @@ -1437,6 +1385,8 @@ static int pm860x_codec_probe(struct platform_device *pdev) pm860x->chip = chip; pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client : chip->companion; + pm860x->regmap = (chip->id == CHIP_PM8607) ? chip->regmap + : chip->regmap_companion; platform_set_drvdata(pdev, pm860x); for (i = 0; i < 4; i++) { diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h index 3364ba4a3607..f7282f4f4a79 100644 --- a/sound/soc/codecs/88pm860x-codec.h +++ b/sound/soc/codecs/88pm860x-codec.h @@ -12,67 +12,66 @@ #ifndef __88PM860X_H #define __88PM860X_H -/* The offset of these registers are 0xb0 */ -#define PM860X_PCM_IFACE_1 0x00 -#define PM860X_PCM_IFACE_2 0x01 -#define PM860X_PCM_IFACE_3 0x02 -#define PM860X_PCM_RATE 0x03 -#define PM860X_EC_PATH 0x04 -#define PM860X_SIDETONE_L_GAIN 0x05 -#define PM860X_SIDETONE_R_GAIN 0x06 -#define PM860X_SIDETONE_SHIFT 0x07 -#define PM860X_ADC_OFFSET_1 0x08 -#define PM860X_ADC_OFFSET_2 0x09 -#define PM860X_DMIC_DELAY 0x0a +#define PM860X_PCM_IFACE_1 0xb0 +#define PM860X_PCM_IFACE_2 0xb1 +#define PM860X_PCM_IFACE_3 0xb2 +#define PM860X_PCM_RATE 0xb3 +#define PM860X_EC_PATH 0xb4 +#define PM860X_SIDETONE_L_GAIN 0xb5 +#define PM860X_SIDETONE_R_GAIN 0xb6 +#define PM860X_SIDETONE_SHIFT 0xb7 +#define PM860X_ADC_OFFSET_1 0xb8 +#define PM860X_ADC_OFFSET_2 0xb9 +#define PM860X_DMIC_DELAY 0xba -#define PM860X_I2S_IFACE_1 0x0b -#define PM860X_I2S_IFACE_2 0x0c -#define PM860X_I2S_IFACE_3 0x0d -#define PM860X_I2S_IFACE_4 0x0e -#define PM860X_EQUALIZER_N0_1 0x0f -#define PM860X_EQUALIZER_N0_2 0x10 -#define PM860X_EQUALIZER_N1_1 0x11 -#define PM860X_EQUALIZER_N1_2 0x12 -#define PM860X_EQUALIZER_D1_1 0x13 -#define PM860X_EQUALIZER_D1_2 0x14 -#define PM860X_LOFI_GAIN_LEFT 0x15 -#define PM860X_LOFI_GAIN_RIGHT 0x16 -#define PM860X_HIFIL_GAIN_LEFT 0x17 -#define PM860X_HIFIL_GAIN_RIGHT 0x18 -#define PM860X_HIFIR_GAIN_LEFT 0x19 -#define PM860X_HIFIR_GAIN_RIGHT 0x1a -#define PM860X_DAC_OFFSET 0x1b -#define PM860X_OFFSET_LEFT_1 0x1c -#define PM860X_OFFSET_LEFT_2 0x1d -#define PM860X_OFFSET_RIGHT_1 0x1e -#define PM860X_OFFSET_RIGHT_2 0x1f -#define PM860X_ADC_ANA_1 0x20 -#define PM860X_ADC_ANA_2 0x21 -#define PM860X_ADC_ANA_3 0x22 -#define PM860X_ADC_ANA_4 0x23 -#define PM860X_ANA_TO_ANA 0x24 -#define PM860X_HS1_CTRL 0x25 -#define PM860X_HS2_CTRL 0x26 -#define PM860X_LO1_CTRL 0x27 -#define PM860X_LO2_CTRL 0x28 -#define PM860X_EAR_CTRL_1 0x29 -#define PM860X_EAR_CTRL_2 0x2a -#define PM860X_AUDIO_SUPPLIES_1 0x2b -#define PM860X_AUDIO_SUPPLIES_2 0x2c -#define PM860X_ADC_EN_1 0x2d -#define PM860X_ADC_EN_2 0x2e -#define PM860X_DAC_EN_1 0x2f -#define PM860X_DAC_EN_2 0x31 -#define PM860X_AUDIO_CAL_1 0x32 -#define PM860X_AUDIO_CAL_2 0x33 -#define PM860X_AUDIO_CAL_3 0x34 -#define PM860X_AUDIO_CAL_4 0x35 -#define PM860X_AUDIO_CAL_5 0x36 -#define PM860X_ANA_INPUT_SEL_1 0x37 -#define PM860X_ANA_INPUT_SEL_2 0x38 +#define PM860X_I2S_IFACE_1 0xbb +#define PM860X_I2S_IFACE_2 0xbc +#define PM860X_I2S_IFACE_3 0xbd +#define PM860X_I2S_IFACE_4 0xbe +#define PM860X_EQUALIZER_N0_1 0xbf +#define PM860X_EQUALIZER_N0_2 0xc0 +#define PM860X_EQUALIZER_N1_1 0xc1 +#define PM860X_EQUALIZER_N1_2 0xc2 +#define PM860X_EQUALIZER_D1_1 0xc3 +#define PM860X_EQUALIZER_D1_2 0xc4 +#define PM860X_LOFI_GAIN_LEFT 0xc5 +#define PM860X_LOFI_GAIN_RIGHT 0xc6 +#define PM860X_HIFIL_GAIN_LEFT 0xc7 +#define PM860X_HIFIL_GAIN_RIGHT 0xc8 +#define PM860X_HIFIR_GAIN_LEFT 0xc9 +#define PM860X_HIFIR_GAIN_RIGHT 0xca +#define PM860X_DAC_OFFSET 0xcb +#define PM860X_OFFSET_LEFT_1 0xcc +#define PM860X_OFFSET_LEFT_2 0xcd +#define PM860X_OFFSET_RIGHT_1 0xce +#define PM860X_OFFSET_RIGHT_2 0xcf +#define PM860X_ADC_ANA_1 0xd0 +#define PM860X_ADC_ANA_2 0xd1 +#define PM860X_ADC_ANA_3 0xd2 +#define PM860X_ADC_ANA_4 0xd3 +#define PM860X_ANA_TO_ANA 0xd4 +#define PM860X_HS1_CTRL 0xd5 +#define PM860X_HS2_CTRL 0xd6 +#define PM860X_LO1_CTRL 0xd7 +#define PM860X_LO2_CTRL 0xd8 +#define PM860X_EAR_CTRL_1 0xd9 +#define PM860X_EAR_CTRL_2 0xda +#define PM860X_AUDIO_SUPPLIES_1 0xdb +#define PM860X_AUDIO_SUPPLIES_2 0xdc +#define PM860X_ADC_EN_1 0xdd +#define PM860X_ADC_EN_2 0xde +#define PM860X_DAC_EN_1 0xdf +#define PM860X_DAC_EN_2 0xe1 +#define PM860X_AUDIO_CAL_1 0xe2 +#define PM860X_AUDIO_CAL_2 0xe3 +#define PM860X_AUDIO_CAL_3 0xe4 +#define PM860X_AUDIO_CAL_4 0xe5 +#define PM860X_AUDIO_CAL_5 0xe6 +#define PM860X_ANA_INPUT_SEL_1 0xe7 +#define PM860X_ANA_INPUT_SEL_2 0xe8 -#define PM860X_PCM_IFACE_4 0x39 -#define PM860X_I2S_IFACE_5 0x3a +#define PM860X_PCM_IFACE_4 0xe9 +#define PM860X_I2S_IFACE_5 0xea #define PM860X_SHORTS 0x3b #define PM860X_PLL_ADJ_1 0x3c -- cgit From 38bfd48b87c44f6958f75bfcd5ae5a53bd3ca07b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 19:17:14 +0100 Subject: ASoC: ab8500: Downgrade noisy log message Signed-off-by: Mark Brown Acked-by: Lee Jones --- sound/soc/codecs/ab8500-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0adacfce..7cea5a8487d0 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2601,7 +2601,7 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) static int ab8500_codec_driver_remove(struct platform_device *pdev) { - dev_info(&pdev->dev, "%s Enter.\n", __func__); + dev_dbg(&pdev->dev, "%s Enter.\n", __func__); snd_soc_unregister_codec(&pdev->dev); -- cgit From 51f20e4cd83e804fb4fd940873763f29616f12a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 19:25:18 +0100 Subject: ASoC: ab8500: Use ASoC I/O functions In preparation for moving away from implementing the ASoC level register I/O functionality change direct calls to the ab8500 implementation of that to use snd_soc_write() Signed-off-by: Mark Brown Acked-by: Lee Jones --- sound/soc/codecs/ab8500-codec.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 7cea5a8487d0..c2b663696611 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2527,12 +2527,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) } /* Override HW-defaults */ - ab8500_codec_write_reg(codec, - AB8500_ANACONF5, - BIT(AB8500_ANACONF5_HSAUTOEN)); - ab8500_codec_write_reg(codec, - AB8500_SHORTCIRCONF, - BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); + snd_soc_write(codec, AB8500_ANACONF5, + BIT(AB8500_ANACONF5_HSAUTOEN)); + snd_soc_write(codec, AB8500_SHORTCIRCONF, + BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); /* Add filter controls */ status = snd_soc_add_codec_controls(codec, ab8500_filter_controls, -- cgit From ff795d614bfa62a3c6fc0bcb75cb8842e5a87892 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 20 Sep 2013 10:38:16 +0100 Subject: ASoC: ab8500: Convert register I/O to regmap As part of a general push to eliminate the duplicated register I/O support in ASoC convert ab8500 to use regmap. Signed-off-by: Mark Brown Acked-by: Lee Jones --- sound/soc/codecs/ab8500-codec.c | 64 +++++++++++++++++++---------------------- 1 file changed, 30 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index c2b663696611..d5a0fc4b2fe2 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -126,6 +126,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -166,49 +168,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; + struct device *dev = context; - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); - - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -2483,6 +2471,8 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); + codec->control_data = drvdata->regmap; + if (np) { if (!pdata) pdata = devm_kzalloc(dev, @@ -2560,9 +2550,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2585,6 +2572,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, -- cgit From 84aac6c79bfdcfbcd8541c814b365c3001cdf5e6 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Sat, 21 Sep 2013 12:00:36 +0200 Subject: ASoC: kirkwood: fix loss of external clock at probe time At probe time, when the clock driver is not yet initialized, the external clock of the kirkwood sound device will not be usable. This patch fixes this problem defering the device probe. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d4ef48..3e59af983527 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -496,7 +496,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return err; priv->extclk = devm_clk_get(&pdev->dev, "extclk"); - if (!IS_ERR(priv->extclk)) { + if (IS_ERR(priv->extclk)) { + if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { if (priv->extclk == priv->clk) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); -- cgit From da5feefeda496f003a2e909762e72843cb9837a6 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 20 Sep 2013 18:19:05 +0100 Subject: ASoC: Docs: Update codec documentation Update the codec class driver documentation and bring it up to date with the current code base. This includes API changes, regmap and multi component. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/codec.txt | 46 +++++++++++----------------------- 1 file changed, 15 insertions(+), 31 deletions(-) diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index bce23a4a7875..db5f9c9ae149 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -1,22 +1,23 @@ -ASoC Codec Driver -================= +ASoC Codec Class Driver +======================= -The codec driver is generic and hardware independent code that configures the -codec to provide audio capture and playback. It should contain no code that is -specific to the target platform or machine. All platform and machine specific -code should be added to the platform and machine drivers respectively. +The codec class driver is generic and hardware independent code that configures +the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. +It should contain no code that is specific to the target platform or machine. +All platform and machine specific code should be added to the platform and +machine drivers respectively. -Each codec driver *must* provide the following features:- +Each codec class driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs + 2) Codec control IO - using RegMap API 3) Mixers and audio controls 4) Codec audio operations + 5) DAPM description. + 6) DAPM event handler. Optionally, codec drivers can also provide:- - 5) DAPM description. - 6) DAPM event handler. 7) DAC Digital mute control. Its probably best to use this guide in conjunction with the existing codec @@ -64,26 +65,9 @@ struct snd_soc_dai_driver wm8731_dai = { 2 - Codec control IO -------------------- The codec can usually be controlled via an I2C or SPI style interface -(AC97 combines control with data in the DAI). The codec drivers provide -functions to read and write the codec registers along with supplying a -register cache:- - - /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; - -Codec read/write should do any data formatting and call the hardware -read write below to perform the IO. These functions are called by the -core and ALSA when performing DAPM or changing the mixer:- - - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - -Codec hardware IO functions - usually points to either the I2C, SPI or AC97 -read/write:- - - hw_write_t hw_write; - hw_read_t hw_read; +(AC97 combines control with data in the DAI). The codec driver should use the +Regmap API for all codec IO. Please see include/linux/regmap.h and existing +codec drivers for example regmap usage. 3 - Mixers and audio controls @@ -127,7 +111,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following ALSA operations:- +The codec driver also supports the following ALSA PCM operations:- /* SoC audio ops */ struct snd_soc_ops { -- cgit From 3eb012834b28585a37fe0684182828efc1ab0512 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 20 Sep 2013 18:19:06 +0100 Subject: ASoC: Docs: Platform update Update the platform class driver documentation and bring it up to date with the current code base. This includes multi component and DSP. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/platform.txt | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index d57efad37e0a..3a08a2c9150c 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -1,9 +1,9 @@ ASoC Platform Driver ==================== -An ASoC platform driver can be divided into audio DMA and SoC DAI configuration -and control. The platform drivers only target the SoC CPU and must have no board -specific code. +An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI +drivers and DSP drivers. The platform drivers only target the SoC CPU and must +have no board specific code. Audio DMA ========= @@ -64,3 +64,16 @@ Each SoC DAI driver must provide the following features:- 5) Suspend and resume (optional) Please see codec.txt for a description of items 1 - 4. + + +SoC DSP Drivers +=============== + +Each SoC DSP driver usually supplies the following features :- + + 1) DAPM graph + 2) Mixer controls + 3) DMA IO to/from DSP buffers (if applicable) + 4) Definition of DSP front end (FE) PCM devices. + +Please see DPCM.txt for a description of item 4. -- cgit From 630a1b3b4f0bb40c84c1fde4fec0d97de62ccdac Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 20 Sep 2013 18:19:07 +0100 Subject: ASoC: Docs: update DAPM Update the DAPM documentation and bring it up to date with the current code base. This includes API changes and new widgets. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/dapm.txt | 71 ++++++++++++++++++++--------------- 1 file changed, 41 insertions(+), 30 deletions(-) diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 05bf5a0eee41..7dfd88ce31ac 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -21,7 +21,7 @@ level power systems. There are 4 power domains within DAPM - 1. Codec domain - VREF, VMID (core codec and audio power) + 1. Codec bias domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. @@ -63,14 +63,22 @@ Audio DAPM widgets fall into a number of types:- o Line - Line Input/Output (and optional Jack) o Speaker - Speaker o Supply - Power or clock supply widget used by other widgets. + o Regulator - External regulator that supplies power to audio components. + o Clock - External clock that supplies clock to audio componnents. + o AIF IN - Audio Interface Input (with TDM slot mask). + o AIF OUT - Audio Interface Output (with TDM slot mask). + o Siggen - Signal Generator. + o DAI IN - Digital Audio Interface Input. + o DAI OUT - Digital Audio Interface Output. + o DAI Link - DAI Link between two DAI structures */ o Pre - Special PRE widget (exec before all others) o Post - Special POST widget (exec after all others) (Widgets are defined in include/sound/soc-dapm.h) -Widgets are usually added in the codec driver and the machine driver. There are -convenience macros defined in soc-dapm.h that can be used to quickly build a -list of widgets of the codecs and machines DAPM widgets. +Widgets can be added to the sound card by any of the component driver types. +There are convenience macros defined in soc-dapm.h that can be used to quickly +build a list of widgets of the codecs and machines DAPM widgets. Most widgets have a name, register, shift and invert. Some widgets have extra parameters for stream name and kcontrols. @@ -80,11 +88,13 @@ parameters for stream name and kcontrols. ------------------------- Stream Widgets relate to the stream power domain and only consist of ADCs -(analog to digital converters) and DACs (digital to analog converters). +(analog to digital converters), DACs (digital to analog converters), +AIF IN and AIF OUT. Stream widgets have the following format:- SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), +SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert) NOTE: the stream name must match the corresponding stream name in your codec snd_soc_codec_dai. @@ -94,6 +104,11 @@ e.g. stream widgets for HiFi playback and capture SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), +e.g. stream widgets for AIF + +SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + 2.2 Path Domain Widgets ----------------------- @@ -121,12 +136,14 @@ If you dont want the mixer elements prefixed with the name of the mixer widget, you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same as for SND_SOC_DAPM_MIXER. -2.3 Platform/Machine domain Widgets ------------------------------------ + +2.3 Machine domain Widgets +-------------------------- Machine widgets are different from codec widgets in that they don't have a codec register bit associated with them. A machine widget is assigned to each -machine audio component (non codec) that can be independently powered. e.g. +machine audio component (non codec or DSP) that can be independently +powered. e.g. o Speaker Amp o Microphone Bias @@ -146,12 +163,12 @@ static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), -2.4 Codec Domain ----------------- +2.4 Codec (BIAS) Domain +----------------------- -The codec power domain has no widgets and is handled by the codecs DAPM event -handler. This handler is called when the codec powerstate is changed wrt to any -stream event or by kernel PM events. +The codec bias power domain has no widgets and is handled by the codecs DAPM +event handler. This handler is called when the codec powerstate is changed wrt +to any stream event or by kernel PM events. 2.5 Virtual Widgets @@ -169,15 +186,16 @@ After all the widgets have been defined, they can then be added to the DAPM subsystem individually with a call to snd_soc_dapm_new_control(). -3. Codec Widget Interconnections -================================ +3. Codec/DSP Widget Interconnections +==================================== -Widgets are connected to each other within the codec and machine by audio paths -(called interconnections). Each interconnection must be defined in order to -create a map of all audio paths between widgets. +Widgets are connected to each other within the codec, platform and machine by +audio paths (called interconnections). Each interconnection must be defined in +order to create a map of all audio paths between widgets. -This is easiest with a diagram of the codec (and schematic of the machine audio -system), as it requires joining widgets together via their audio signal paths. +This is easiest with a diagram of the codec or DSP (and schematic of the machine +audio system), as it requires joining widgets together via their audio signal +paths. e.g., from the WM8731 output mixer (wm8731.c) @@ -247,16 +265,9 @@ machine and includes the codec. e.g. o Mic Jack o Codec Pins -When a codec pin is NC it can be marked as not used with a call to - -snd_soc_dapm_set_endpoint(codec, "Widget Name", 0); - -The last argument is 0 for inactive and 1 for active. This way the pin and its -input widget will never be powered up and consume power. - -This also applies to machine widgets. e.g. if a headphone is connected to a -jack then the jack can be marked active. If the headphone is removed, then -the headphone jack can be marked inactive. +Endpoints are added to the DAPM graph so that their usage can be determined in +order to save power. e.g. NC codecs pins will be switched OFF, unconnected +jacks can also be switched OFF. 5 DAPM Widget Events -- cgit From b5c47df974ddae44d4a1ff935cdda30b0795bc00 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 20 Sep 2013 18:19:08 +0100 Subject: ASoC: Docs: Machine update Update the machine driver documentation and bring it up to date with the current code base. This includes multi component. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/machine.txt | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index d50c14df3411..74056dba52be 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -1,8 +1,10 @@ ASoC Machine Driver =================== -The ASoC machine (or board) driver is the code that glues together the platform -and codec drivers. +The ASoC machine (or board) driver is the code that glues together all the +component drivers (e.g. codecs, platforms and DAIs). It also describes the +relationships between each componnent which include audio paths, GPIOs, +interrupts, clocking, jacks and voltage regulators. The machine driver can contain codec and platform specific code. It registers the audio subsystem with the kernel as a platform device and is represented by -- cgit From 469b7bc4e6dbfdb173f0901f746e9277f6740ba7 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 20 Sep 2013 18:19:09 +0100 Subject: ASoC: Docs: Add documentation for Dynamic PCM Add documentation describing DPCM with examples of a DSP based smart phone. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/DPCM.txt | 380 ++++++++++++++++++++++++++++++++++ 1 file changed, 380 insertions(+) create mode 100644 Documentation/sound/alsa/soc/DPCM.txt diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt new file mode 100644 index 000000000000..aa8546f2d144 --- /dev/null +++ b/Documentation/sound/alsa/soc/DPCM.txt @@ -0,0 +1,380 @@ +Dynamic PCM +=========== + +1. Description +============== + +Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to +various digital endpoints during the PCM stream runtime. e.g. PCM0 can route +digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP +drivers that expose several ALSA PCMs and can route to multiple DAIs. + +The DPCM runtime routing is determined by the ALSA mixer settings in the same +way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM +graph representing the DSP internal audio paths and uses the mixer settings to +determine the patch used by each ALSA PCM. + +DPCM re-uses all the existing component codec, platform and DAI drivers without +any modifications. + + +Phone Audio System with SoC based DSP +------------------------------------- + +Consider the following phone audio subsystem. This will be used in this +document for all examples :- + +| Front End PCMs | SoC DSP | Back End DAIs | Audio devices | + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +This diagram shows a simple smart phone audio subsystem. It supports Bluetooth, +FM digital radio, Speakers, Headset Jack, digital microphones and cellular +modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and +supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any +of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI. + + + +Example - DPCM Switching playback from DAI0 to DAI1 +--------------------------------------------------- + +Audio is being played to the Headset. After a while the user removes the headset +and audio continues playing on the speakers. + +Playback on PCM0 to Headset would look like :- + + ************* +PCM0 <============> * * <====DAI0=====> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +The headset is removed from the jack by user so the speakers must now be used :- + + ************* +PCM0 <============> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <====DAI1=====> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +The audio driver processes this as follows :- + + 1) Machine driver receives Jack removal event. + + 2) Machine driver OR audio HAL disables the Headset path. + + 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0 + for headset since the path is now disabled. + + 4) Machine driver or audio HAL enables the speaker path. + + 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and + trigger(start) for DAI1 Speakers since the path is enabled. + +In this example, the machine driver or userspace audio HAL can alter the routing +and then DPCM will take care of managing the DAI PCM operations to either bring +the link up or down. Audio playback does not stop during this transition. + + + +DPCM machine driver +=================== + +The DPCM enabled ASoC machine driver is similar to normal machine drivers +except that we also have to :- + + 1) Define the FE and BE DAI links. + + 2) Define any FE/BE PCM operations. + + 3) Define widget graph connections. + + +1 FE and BE DAI links +--------------------- + +| Front End PCMs | SoC DSP | Back End DAIs | Audio devices | + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +For the example above we have to define 4 FE DAI links and 6 BE DAI links. The +FE DAI links are defined as follows :- + +static struct snd_soc_dai_link machine_dais[] = { + { + .name = "PCM0 System", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "dsp-audio", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + .....< other FE and BE DAI links here > +}; + +This FE DAI link is pretty similar to a regular DAI link except that we also +set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream +directions should also be set with the "dpcm_playback" and "dpcm_capture" +flags. There is also an option to specify the ordering of the trigger call for +each FE. This allows the ASoC core to trigger the DSP before or after the other +components (as some DSPs have strong requirements for the ordering DAI/DSP +start and stop sequences). + +The FE DAI above sets the codec and code DAIs to dummy devices since the BE is +dynamic and will change depending on runtime config. + +The BE DAIs are configured as follows :- + +static struct snd_soc_dai_link machine_dais[] = { + .....< FE DAI links here > + { + .name = "Codec Headset", + .cpu_dai_name = "ssp-dai.0", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "rt5640.0-001c", + .codec_dai_name = "rt5640-aif1", + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = hswult_ssp0_fixup, + .ops = &haswell_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + .....< other BE DAI links here > +}; + +This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets +the "no_pcm" flag to mark it has a BE and sets flags for supported stream +directions using "dpcm_playback" and "dpcm_capture" above. + +The BE has also flags set for ignoreing suspend and PM down time. This allows +the BE to work in a hostless mode where the host CPU is not transferring data +like a BT phone call :- + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <====DAI2=====> MODEM + * * +PCM3 <------------> * * <====DAI3=====> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are +still in operation. + +A BE DAI link can also set the codec to a dummy device if the code is a device +that is managed externally. + +Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the +DSP firmware. + + +2 FE/BE PCM operations +---------------------- + +The BE above also exports some PCM operations and a "fixup" callback. The fixup +callback is used by the machine driver to (re)configure the DAI based upon the +FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE. + +e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for +DAI0. This means all FE hw_params have to be fixed in the machine driver for +DAI0 so that the DAI is running at desired configuration regardless of the FE +configuration. + +static int dai0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set DAI0 to 16 bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +The other PCM operation are the same as for regular DAI links. Use as necessary. + + +3 Widget graph connections +-------------------------- + +The BE DAI links will normally be connected to the graph at initialisation time +by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this +has to be set explicitly in the driver :- + +/* BE for codec Headset - DAI0 is dummy and managed by DSP FW */ +{"DAI0 CODEC IN", NULL, "AIF1 Capture"}, +{"AIF1 Playback", NULL, "DAI0 CODEC OUT"}, + + +Writing a DPCM DSP driver +========================= + +The DPCM DSP driver looks much like a standard platform class ASoC driver +combined with elements from a codec class driver. A DSP platform driver must +implement :- + + 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver. + + 2) DAPM graph showing DSP audio routing from FE DAIs to BEs. + + 3) DAPM widgets from DSP graph. + + 4) Mixers for gains, routing, etc. + + 5) DMA configuration. + + 6) BE AIF widgets. + +Items 6 is important for routing the audio outside of the DSP. AIF need to be +defined for each BE and each stream direction. e.g for BE DAI0 above we would +have :- + +SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0), + +The BE AIF are used to connect the DSP graph to the graphs for the other +component drivers (e.g. codec graph). + + +Hostless PCM streams +==================== + +A hostless PCM stream is a stream that is not routed through the host CPU. An +example of this would be a phone call from handset to modem. + + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <====DAI1=====> Codec Speakers/Mic + * DSP * +PCM2 <------------> * * <====DAI2=====> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +In this case the PCM data is routed via the DSP. The host CPU in this use case +is only used for control and can sleep during the runtime of the stream. + +The host can control the hostless link either by :- + + 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link + is enabled or disabled by the state of the DAPM graph. This usually means + there is a mixer control that can be used to connect or disconnect the path + between both DAIs. + + 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM + graph. Control is then carried out by the FE as regualar PCM operations. + This method gives more control over the DAI links, but requires much more + userspace code to control the link. Its recommended to use CODEC<->CODEC + unless your HW needs more fine grained sequencing of the PCM ops. + + +CODEC <-> CODEC link +-------------------- + +This DAI link is enabled when DAPM detects a valid path within the DAPM graph. +The machine driver sets some additional parameters to the DAI link i.e. + +static const struct snd_soc_pcm_stream dai_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_dai_link dais[] = { + < ... more DAI links above ... > + { + .name = "MODEM", + .stream_name = "MODEM", + .cpu_dai_name = "dai2", + .codec_dai_name = "modem-aif1", + .codec_name = "modem", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .params = &dai_params, + } + < ... more DAI links here ... > + +These parameters are used to configure the DAI hw_params() when DAPM detects a +valid path and then calls the PCM operations to start the link. DAPM will also +call the appropriate PCM operations to disable the DAI when the path is no +longer valid. + + +Hostless FE +----------- + +The DAI link(s) are enabled by a FE that does not read or write any PCM data. +This means creating a new FE that is connected with a virtual path to both +DAI links. The DAI links will be started when the FE PCM is started and stopped +when the FE PCM is stopped. Note that the FE PCM cannot read or write data in +this configuration. + + -- cgit From 64d2307c3b7daa03dbc0c3a6b514709dd7b6eaee Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 23 Sep 2013 01:08:25 -0300 Subject: ASoC: fsl: fsl_ssi: Fix simultaneous capture and playback When doing simultaneous capture and playback on a mx6 board we get the following error: $ arecord -f cd | aplay -f cd imx-sgtl5000 sound.13: set sample size in capture stream first fsl-ssi-dai 2028000.ssi: ASoC: can't open interface 2028000.ssi: -11 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_ open) unable to open slave aplay: main:660: audio open error: Device or resource busy Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo The 'arecord -f cd | aplay -f cd' always trigger cause the 'if (!first_runtime->sample_bits)' block to be true which returns an error. Adjust the logic inside fsl_ssi_startup(), so that we do not always hit the error when playing 'arecord | aplay' line for the first time. Reported-by: Chris Clepper Suggested-by: Nicolin Chen Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6ac87300d45d..cdbb641ef518 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -469,19 +469,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * parameters, then the second stream may be * constrained to the wrong sample rate or size. */ - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == - SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + if (first_runtime->sample_bits) { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } } ssi_private->second_stream = substream; -- cgit From d3be689e6a07c00123786659b4429b07cf4272ac Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 24 Sep 2013 01:25:08 -0700 Subject: ASoC: rcar: remove unnecessary mach/clock.h ${LINUX}/sound/soc/sh driver can be compiled from SuperH and ARM. but, ${LINUX}/sound/soc/sh/rcar driver included SH-ARM specific header. This patch removes it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index d80deb7ccf13..2935bbf1811b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -8,7 +8,6 @@ * for more details. */ #include -#include #include "rsnd.h" #define CLKA 0 -- cgit From 356d86e24850cdc353602b90be73e627f86707c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 17:22:17 +0100 Subject: ASoC: max98088: Fix indentation Tested-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 46 ++++++++++++++++++++++----------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 566a367c94fa..391f66913a44 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -38,20 +38,20 @@ struct max98088_cdata { }; struct max98088_priv { - enum max98088_type devtype; - struct max98088_pdata *pdata; - unsigned int sysclk; - struct max98088_cdata dai[2]; - int eq_textcnt; - const char **eq_texts; - struct soc_enum eq_enum; - u8 ina_state; - u8 inb_state; - unsigned int ex_mode; - unsigned int digmic; - unsigned int mic1pre; - unsigned int mic2pre; - unsigned int extmic_mode; + enum max98088_type devtype; + struct max98088_pdata *pdata; + unsigned int sysclk; + struct max98088_cdata dai[2]; + int eq_textcnt; + const char **eq_texts; + struct soc_enum eq_enum; + u8 ina_state; + u8 inb_state; + unsigned int ex_mode; + unsigned int digmic; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mode; }; static const u8 max98088_reg[M98088_REG_CNT] = { @@ -2066,15 +2066,15 @@ static int max98088_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_max98088 = { - .probe = max98088_probe, - .remove = max98088_remove, - .suspend = max98088_suspend, - .resume = max98088_resume, - .set_bias_level = max98088_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max98088_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98088_reg, - .volatile_register = max98088_volatile_register, + .probe = max98088_probe, + .remove = max98088_remove, + .suspend = max98088_suspend, + .resume = max98088_resume, + .set_bias_level = max98088_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max98088_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max98088_reg, + .volatile_register = max98088_volatile_register, .dapm_widgets = max98088_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), .dapm_routes = max98088_audio_map, -- cgit From ad65adf4a3039ecd93d4712ac6524dbd9e0e848a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 17:54:02 +0100 Subject: ASoC: max98088: Use table based control init Tested-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 391f66913a44..8896d5e33980 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2048,9 +2048,6 @@ static int max98088_probe(struct snd_soc_codec *codec) max98088_handle_pdata(codec); - snd_soc_add_codec_controls(codec, max98088_snd_controls, - ARRAY_SIZE(max98088_snd_controls)); - err_access: return ret; } @@ -2071,6 +2068,8 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { .suspend = max98088_suspend, .resume = max98088_resume, .set_bias_level = max98088_set_bias_level, + .controls = max98088_snd_controls, + .num_controls = ARRAY_SIZE(max98088_snd_controls), .reg_cache_size = ARRAY_SIZE(max98088_reg), .reg_word_size = sizeof(u8), .reg_cache_default = max98088_reg, -- cgit From 4127d5d59f8135e3c187b8daa2540691761938ce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 17:56:17 +0100 Subject: ASoC: max98088: Convert to direct regmap API usage This saves code and moves us towards removing the redundant register I/O implementation in ASoC. Tested-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 580 +++++++++++++++++++------------------------- 1 file changed, 251 insertions(+), 329 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 8896d5e33980..31912d59702c 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include @@ -38,6 +39,7 @@ struct max98088_cdata { }; struct max98088_priv { + struct regmap *regmap; enum max98088_type devtype; struct max98088_pdata *pdata; unsigned int sysclk; @@ -54,278 +56,206 @@ struct max98088_priv { unsigned int extmic_mode; }; -static const u8 max98088_reg[M98088_REG_CNT] = { - 0x00, /* 00 IRQ status */ - 0x00, /* 01 MIC status */ - 0x00, /* 02 jack status */ - 0x00, /* 03 battery voltage */ - 0x00, /* 04 */ - 0x00, /* 05 */ - 0x00, /* 06 */ - 0x00, /* 07 */ - 0x00, /* 08 */ - 0x00, /* 09 */ - 0x00, /* 0A */ - 0x00, /* 0B */ - 0x00, /* 0C */ - 0x00, /* 0D */ - 0x00, /* 0E */ - 0x00, /* 0F interrupt enable */ - - 0x00, /* 10 master clock */ - 0x00, /* 11 DAI1 clock mode */ - 0x00, /* 12 DAI1 clock control */ - 0x00, /* 13 DAI1 clock control */ - 0x00, /* 14 DAI1 format */ - 0x00, /* 15 DAI1 clock */ - 0x00, /* 16 DAI1 config */ - 0x00, /* 17 DAI1 TDM */ - 0x00, /* 18 DAI1 filters */ - 0x00, /* 19 DAI2 clock mode */ - 0x00, /* 1A DAI2 clock control */ - 0x00, /* 1B DAI2 clock control */ - 0x00, /* 1C DAI2 format */ - 0x00, /* 1D DAI2 clock */ - 0x00, /* 1E DAI2 config */ - 0x00, /* 1F DAI2 TDM */ - - 0x00, /* 20 DAI2 filters */ - 0x00, /* 21 data config */ - 0x00, /* 22 DAC mixer */ - 0x00, /* 23 left ADC mixer */ - 0x00, /* 24 right ADC mixer */ - 0x00, /* 25 left HP mixer */ - 0x00, /* 26 right HP mixer */ - 0x00, /* 27 HP control */ - 0x00, /* 28 left REC mixer */ - 0x00, /* 29 right REC mixer */ - 0x00, /* 2A REC control */ - 0x00, /* 2B left SPK mixer */ - 0x00, /* 2C right SPK mixer */ - 0x00, /* 2D SPK control */ - 0x00, /* 2E sidetone */ - 0x00, /* 2F DAI1 playback level */ - - 0x00, /* 30 DAI1 playback level */ - 0x00, /* 31 DAI2 playback level */ - 0x00, /* 32 DAI2 playbakc level */ - 0x00, /* 33 left ADC level */ - 0x00, /* 34 right ADC level */ - 0x00, /* 35 MIC1 level */ - 0x00, /* 36 MIC2 level */ - 0x00, /* 37 INA level */ - 0x00, /* 38 INB level */ - 0x00, /* 39 left HP volume */ - 0x00, /* 3A right HP volume */ - 0x00, /* 3B left REC volume */ - 0x00, /* 3C right REC volume */ - 0x00, /* 3D left SPK volume */ - 0x00, /* 3E right SPK volume */ - 0x00, /* 3F MIC config */ - - 0x00, /* 40 MIC threshold */ - 0x00, /* 41 excursion limiter filter */ - 0x00, /* 42 excursion limiter threshold */ - 0x00, /* 43 ALC */ - 0x00, /* 44 power limiter threshold */ - 0x00, /* 45 power limiter config */ - 0x00, /* 46 distortion limiter config */ - 0x00, /* 47 audio input */ - 0x00, /* 48 microphone */ - 0x00, /* 49 level control */ - 0x00, /* 4A bypass switches */ - 0x00, /* 4B jack detect */ - 0x00, /* 4C input enable */ - 0x00, /* 4D output enable */ - 0xF0, /* 4E bias control */ - 0x00, /* 4F DAC power */ - - 0x0F, /* 50 DAC power */ - 0x00, /* 51 system */ - 0x00, /* 52 DAI1 EQ1 */ - 0x00, /* 53 DAI1 EQ1 */ - 0x00, /* 54 DAI1 EQ1 */ - 0x00, /* 55 DAI1 EQ1 */ - 0x00, /* 56 DAI1 EQ1 */ - 0x00, /* 57 DAI1 EQ1 */ - 0x00, /* 58 DAI1 EQ1 */ - 0x00, /* 59 DAI1 EQ1 */ - 0x00, /* 5A DAI1 EQ1 */ - 0x00, /* 5B DAI1 EQ1 */ - 0x00, /* 5C DAI1 EQ2 */ - 0x00, /* 5D DAI1 EQ2 */ - 0x00, /* 5E DAI1 EQ2 */ - 0x00, /* 5F DAI1 EQ2 */ - - 0x00, /* 60 DAI1 EQ2 */ - 0x00, /* 61 DAI1 EQ2 */ - 0x00, /* 62 DAI1 EQ2 */ - 0x00, /* 63 DAI1 EQ2 */ - 0x00, /* 64 DAI1 EQ2 */ - 0x00, /* 65 DAI1 EQ2 */ - 0x00, /* 66 DAI1 EQ3 */ - 0x00, /* 67 DAI1 EQ3 */ - 0x00, /* 68 DAI1 EQ3 */ - 0x00, /* 69 DAI1 EQ3 */ - 0x00, /* 6A DAI1 EQ3 */ - 0x00, /* 6B DAI1 EQ3 */ - 0x00, /* 6C DAI1 EQ3 */ - 0x00, /* 6D DAI1 EQ3 */ - 0x00, /* 6E DAI1 EQ3 */ - 0x00, /* 6F DAI1 EQ3 */ - - 0x00, /* 70 DAI1 EQ4 */ - 0x00, /* 71 DAI1 EQ4 */ - 0x00, /* 72 DAI1 EQ4 */ - 0x00, /* 73 DAI1 EQ4 */ - 0x00, /* 74 DAI1 EQ4 */ - 0x00, /* 75 DAI1 EQ4 */ - 0x00, /* 76 DAI1 EQ4 */ - 0x00, /* 77 DAI1 EQ4 */ - 0x00, /* 78 DAI1 EQ4 */ - 0x00, /* 79 DAI1 EQ4 */ - 0x00, /* 7A DAI1 EQ5 */ - 0x00, /* 7B DAI1 EQ5 */ - 0x00, /* 7C DAI1 EQ5 */ - 0x00, /* 7D DAI1 EQ5 */ - 0x00, /* 7E DAI1 EQ5 */ - 0x00, /* 7F DAI1 EQ5 */ - - 0x00, /* 80 DAI1 EQ5 */ - 0x00, /* 81 DAI1 EQ5 */ - 0x00, /* 82 DAI1 EQ5 */ - 0x00, /* 83 DAI1 EQ5 */ - 0x00, /* 84 DAI2 EQ1 */ - 0x00, /* 85 DAI2 EQ1 */ - 0x00, /* 86 DAI2 EQ1 */ - 0x00, /* 87 DAI2 EQ1 */ - 0x00, /* 88 DAI2 EQ1 */ - 0x00, /* 89 DAI2 EQ1 */ - 0x00, /* 8A DAI2 EQ1 */ - 0x00, /* 8B DAI2 EQ1 */ - 0x00, /* 8C DAI2 EQ1 */ - 0x00, /* 8D DAI2 EQ1 */ - 0x00, /* 8E DAI2 EQ2 */ - 0x00, /* 8F DAI2 EQ2 */ - - 0x00, /* 90 DAI2 EQ2 */ - 0x00, /* 91 DAI2 EQ2 */ - 0x00, /* 92 DAI2 EQ2 */ - 0x00, /* 93 DAI2 EQ2 */ - 0x00, /* 94 DAI2 EQ2 */ - 0x00, /* 95 DAI2 EQ2 */ - 0x00, /* 96 DAI2 EQ2 */ - 0x00, /* 97 DAI2 EQ2 */ - 0x00, /* 98 DAI2 EQ3 */ - 0x00, /* 99 DAI2 EQ3 */ - 0x00, /* 9A DAI2 EQ3 */ - 0x00, /* 9B DAI2 EQ3 */ - 0x00, /* 9C DAI2 EQ3 */ - 0x00, /* 9D DAI2 EQ3 */ - 0x00, /* 9E DAI2 EQ3 */ - 0x00, /* 9F DAI2 EQ3 */ - - 0x00, /* A0 DAI2 EQ3 */ - 0x00, /* A1 DAI2 EQ3 */ - 0x00, /* A2 DAI2 EQ4 */ - 0x00, /* A3 DAI2 EQ4 */ - 0x00, /* A4 DAI2 EQ4 */ - 0x00, /* A5 DAI2 EQ4 */ - 0x00, /* A6 DAI2 EQ4 */ - 0x00, /* A7 DAI2 EQ4 */ - 0x00, /* A8 DAI2 EQ4 */ - 0x00, /* A9 DAI2 EQ4 */ - 0x00, /* AA DAI2 EQ4 */ - 0x00, /* AB DAI2 EQ4 */ - 0x00, /* AC DAI2 EQ5 */ - 0x00, /* AD DAI2 EQ5 */ - 0x00, /* AE DAI2 EQ5 */ - 0x00, /* AF DAI2 EQ5 */ - - 0x00, /* B0 DAI2 EQ5 */ - 0x00, /* B1 DAI2 EQ5 */ - 0x00, /* B2 DAI2 EQ5 */ - 0x00, /* B3 DAI2 EQ5 */ - 0x00, /* B4 DAI2 EQ5 */ - 0x00, /* B5 DAI2 EQ5 */ - 0x00, /* B6 DAI1 biquad */ - 0x00, /* B7 DAI1 biquad */ - 0x00, /* B8 DAI1 biquad */ - 0x00, /* B9 DAI1 biquad */ - 0x00, /* BA DAI1 biquad */ - 0x00, /* BB DAI1 biquad */ - 0x00, /* BC DAI1 biquad */ - 0x00, /* BD DAI1 biquad */ - 0x00, /* BE DAI1 biquad */ - 0x00, /* BF DAI1 biquad */ - - 0x00, /* C0 DAI2 biquad */ - 0x00, /* C1 DAI2 biquad */ - 0x00, /* C2 DAI2 biquad */ - 0x00, /* C3 DAI2 biquad */ - 0x00, /* C4 DAI2 biquad */ - 0x00, /* C5 DAI2 biquad */ - 0x00, /* C6 DAI2 biquad */ - 0x00, /* C7 DAI2 biquad */ - 0x00, /* C8 DAI2 biquad */ - 0x00, /* C9 DAI2 biquad */ - 0x00, /* CA */ - 0x00, /* CB */ - 0x00, /* CC */ - 0x00, /* CD */ - 0x00, /* CE */ - 0x00, /* CF */ - - 0x00, /* D0 */ - 0x00, /* D1 */ - 0x00, /* D2 */ - 0x00, /* D3 */ - 0x00, /* D4 */ - 0x00, /* D5 */ - 0x00, /* D6 */ - 0x00, /* D7 */ - 0x00, /* D8 */ - 0x00, /* D9 */ - 0x00, /* DA */ - 0x70, /* DB */ - 0x00, /* DC */ - 0x00, /* DD */ - 0x00, /* DE */ - 0x00, /* DF */ - - 0x00, /* E0 */ - 0x00, /* E1 */ - 0x00, /* E2 */ - 0x00, /* E3 */ - 0x00, /* E4 */ - 0x00, /* E5 */ - 0x00, /* E6 */ - 0x00, /* E7 */ - 0x00, /* E8 */ - 0x00, /* E9 */ - 0x00, /* EA */ - 0x00, /* EB */ - 0x00, /* EC */ - 0x00, /* ED */ - 0x00, /* EE */ - 0x00, /* EF */ - - 0x00, /* F0 */ - 0x00, /* F1 */ - 0x00, /* F2 */ - 0x00, /* F3 */ - 0x00, /* F4 */ - 0x00, /* F5 */ - 0x00, /* F6 */ - 0x00, /* F7 */ - 0x00, /* F8 */ - 0x00, /* F9 */ - 0x00, /* FA */ - 0x00, /* FB */ - 0x00, /* FC */ - 0x00, /* FD */ - 0x00, /* FE */ - 0x00, /* FF */ +static const struct reg_default max98088_reg[] = { + { 0xf, 0x00 }, /* 0F interrupt enable */ + + { 0x10, 0x00 }, /* 10 master clock */ + { 0x11, 0x00 }, /* 11 DAI1 clock mode */ + { 0x12, 0x00 }, /* 12 DAI1 clock control */ + { 0x13, 0x00 }, /* 13 DAI1 clock control */ + { 0x14, 0x00 }, /* 14 DAI1 format */ + { 0x15, 0x00 }, /* 15 DAI1 clock */ + { 0x16, 0x00 }, /* 16 DAI1 config */ + { 0x17, 0x00 }, /* 17 DAI1 TDM */ + { 0x18, 0x00 }, /* 18 DAI1 filters */ + { 0x19, 0x00 }, /* 19 DAI2 clock mode */ + { 0x1a, 0x00 }, /* 1A DAI2 clock control */ + { 0x1b, 0x00 }, /* 1B DAI2 clock control */ + { 0x1c, 0x00 }, /* 1C DAI2 format */ + { 0x1d, 0x00 }, /* 1D DAI2 clock */ + { 0x1e, 0x00 }, /* 1E DAI2 config */ + { 0x1f, 0x00 }, /* 1F DAI2 TDM */ + + { 0x20, 0x00 }, /* 20 DAI2 filters */ + { 0x21, 0x00 }, /* 21 data config */ + { 0x22, 0x00 }, /* 22 DAC mixer */ + { 0x23, 0x00 }, /* 23 left ADC mixer */ + { 0x24, 0x00 }, /* 24 right ADC mixer */ + { 0x25, 0x00 }, /* 25 left HP mixer */ + { 0x26, 0x00 }, /* 26 right HP mixer */ + { 0x27, 0x00 }, /* 27 HP control */ + { 0x28, 0x00 }, /* 28 left REC mixer */ + { 0x29, 0x00 }, /* 29 right REC mixer */ + { 0x2a, 0x00 }, /* 2A REC control */ + { 0x2b, 0x00 }, /* 2B left SPK mixer */ + { 0x2c, 0x00 }, /* 2C right SPK mixer */ + { 0x2d, 0x00 }, /* 2D SPK control */ + { 0x2e, 0x00 }, /* 2E sidetone */ + { 0x2f, 0x00 }, /* 2F DAI1 playback level */ + + { 0x30, 0x00 }, /* 30 DAI1 playback level */ + { 0x31, 0x00 }, /* 31 DAI2 playback level */ + { 0x32, 0x00 }, /* 32 DAI2 playbakc level */ + { 0x33, 0x00 }, /* 33 left ADC level */ + { 0x34, 0x00 }, /* 34 right ADC level */ + { 0x35, 0x00 }, /* 35 MIC1 level */ + { 0x36, 0x00 }, /* 36 MIC2 level */ + { 0x37, 0x00 }, /* 37 INA level */ + { 0x38, 0x00 }, /* 38 INB level */ + { 0x39, 0x00 }, /* 39 left HP volume */ + { 0x3a, 0x00 }, /* 3A right HP volume */ + { 0x3b, 0x00 }, /* 3B left REC volume */ + { 0x3c, 0x00 }, /* 3C right REC volume */ + { 0x3d, 0x00 }, /* 3D left SPK volume */ + { 0x3e, 0x00 }, /* 3E right SPK volume */ + { 0x3f, 0x00 }, /* 3F MIC config */ + + { 0x40, 0x00 }, /* 40 MIC threshold */ + { 0x41, 0x00 }, /* 41 excursion limiter filter */ + { 0x42, 0x00 }, /* 42 excursion limiter threshold */ + { 0x43, 0x00 }, /* 43 ALC */ + { 0x44, 0x00 }, /* 44 power limiter threshold */ + { 0x45, 0x00 }, /* 45 power limiter config */ + { 0x46, 0x00 }, /* 46 distortion limiter config */ + { 0x47, 0x00 }, /* 47 audio input */ + { 0x48, 0x00 }, /* 48 microphone */ + { 0x49, 0x00 }, /* 49 level control */ + { 0x4a, 0x00 }, /* 4A bypass switches */ + { 0x4b, 0x00 }, /* 4B jack detect */ + { 0x4c, 0x00 }, /* 4C input enable */ + { 0x4d, 0x00 }, /* 4D output enable */ + { 0x4e, 0xF0 }, /* 4E bias control */ + { 0x4f, 0x00 }, /* 4F DAC power */ + + { 0x50, 0x0F }, /* 50 DAC power */ + { 0x51, 0x00 }, /* 51 system */ + { 0x52, 0x00 }, /* 52 DAI1 EQ1 */ + { 0x53, 0x00 }, /* 53 DAI1 EQ1 */ + { 0x54, 0x00 }, /* 54 DAI1 EQ1 */ + { 0x55, 0x00 }, /* 55 DAI1 EQ1 */ + { 0x56, 0x00 }, /* 56 DAI1 EQ1 */ + { 0x57, 0x00 }, /* 57 DAI1 EQ1 */ + { 0x58, 0x00 }, /* 58 DAI1 EQ1 */ + { 0x59, 0x00 }, /* 59 DAI1 EQ1 */ + { 0x5a, 0x00 }, /* 5A DAI1 EQ1 */ + { 0x5b, 0x00 }, /* 5B DAI1 EQ1 */ + { 0x5c, 0x00 }, /* 5C DAI1 EQ2 */ + { 0x5d, 0x00 }, /* 5D DAI1 EQ2 */ + { 0x5e, 0x00 }, /* 5E DAI1 EQ2 */ + { 0x5f, 0x00 }, /* 5F DAI1 EQ2 */ + + { 0x60, 0x00 }, /* 60 DAI1 EQ2 */ + { 0x61, 0x00 }, /* 61 DAI1 EQ2 */ + { 0x62, 0x00 }, /* 62 DAI1 EQ2 */ + { 0x63, 0x00 }, /* 63 DAI1 EQ2 */ + { 0x64, 0x00 }, /* 64 DAI1 EQ2 */ + { 0x65, 0x00 }, /* 65 DAI1 EQ2 */ + { 0x66, 0x00 }, /* 66 DAI1 EQ3 */ + { 0x67, 0x00 }, /* 67 DAI1 EQ3 */ + { 0x68, 0x00 }, /* 68 DAI1 EQ3 */ + { 0x69, 0x00 }, /* 69 DAI1 EQ3 */ + { 0x6a, 0x00 }, /* 6A DAI1 EQ3 */ + { 0x6b, 0x00 }, /* 6B DAI1 EQ3 */ + { 0x6c, 0x00 }, /* 6C DAI1 EQ3 */ + { 0x6d, 0x00 }, /* 6D DAI1 EQ3 */ + { 0x6e, 0x00 }, /* 6E DAI1 EQ3 */ + { 0x6f, 0x00 }, /* 6F DAI1 EQ3 */ + + { 0x70, 0x00 }, /* 70 DAI1 EQ4 */ + { 0x71, 0x00 }, /* 71 DAI1 EQ4 */ + { 0x72, 0x00 }, /* 72 DAI1 EQ4 */ + { 0x73, 0x00 }, /* 73 DAI1 EQ4 */ + { 0x74, 0x00 }, /* 74 DAI1 EQ4 */ + { 0x75, 0x00 }, /* 75 DAI1 EQ4 */ + { 0x76, 0x00 }, /* 76 DAI1 EQ4 */ + { 0x77, 0x00 }, /* 77 DAI1 EQ4 */ + { 0x78, 0x00 }, /* 78 DAI1 EQ4 */ + { 0x79, 0x00 }, /* 79 DAI1 EQ4 */ + { 0x7a, 0x00 }, /* 7A DAI1 EQ5 */ + { 0x7b, 0x00 }, /* 7B DAI1 EQ5 */ + { 0x7c, 0x00 }, /* 7C DAI1 EQ5 */ + { 0x7d, 0x00 }, /* 7D DAI1 EQ5 */ + { 0x7e, 0x00 }, /* 7E DAI1 EQ5 */ + { 0x7f, 0x00 }, /* 7F DAI1 EQ5 */ + + { 0x80, 0x00 }, /* 80 DAI1 EQ5 */ + { 0x81, 0x00 }, /* 81 DAI1 EQ5 */ + { 0x82, 0x00 }, /* 82 DAI1 EQ5 */ + { 0x83, 0x00 }, /* 83 DAI1 EQ5 */ + { 0x84, 0x00 }, /* 84 DAI2 EQ1 */ + { 0x85, 0x00 }, /* 85 DAI2 EQ1 */ + { 0x86, 0x00 }, /* 86 DAI2 EQ1 */ + { 0x87, 0x00 }, /* 87 DAI2 EQ1 */ + { 0x88, 0x00 }, /* 88 DAI2 EQ1 */ + { 0x89, 0x00 }, /* 89 DAI2 EQ1 */ + { 0x8a, 0x00 }, /* 8A DAI2 EQ1 */ + { 0x8b, 0x00 }, /* 8B DAI2 EQ1 */ + { 0x8c, 0x00 }, /* 8C DAI2 EQ1 */ + { 0x8d, 0x00 }, /* 8D DAI2 EQ1 */ + { 0x8e, 0x00 }, /* 8E DAI2 EQ2 */ + { 0x8f, 0x00 }, /* 8F DAI2 EQ2 */ + + { 0x90, 0x00 }, /* 90 DAI2 EQ2 */ + { 0x91, 0x00 }, /* 91 DAI2 EQ2 */ + { 0x92, 0x00 }, /* 92 DAI2 EQ2 */ + { 0x93, 0x00 }, /* 93 DAI2 EQ2 */ + { 0x94, 0x00 }, /* 94 DAI2 EQ2 */ + { 0x95, 0x00 }, /* 95 DAI2 EQ2 */ + { 0x96, 0x00 }, /* 96 DAI2 EQ2 */ + { 0x97, 0x00 }, /* 97 DAI2 EQ2 */ + { 0x98, 0x00 }, /* 98 DAI2 EQ3 */ + { 0x99, 0x00 }, /* 99 DAI2 EQ3 */ + { 0x9a, 0x00 }, /* 9A DAI2 EQ3 */ + { 0x9b, 0x00 }, /* 9B DAI2 EQ3 */ + { 0x9c, 0x00 }, /* 9C DAI2 EQ3 */ + { 0x9d, 0x00 }, /* 9D DAI2 EQ3 */ + { 0x9e, 0x00 }, /* 9E DAI2 EQ3 */ + { 0x9f, 0x00 }, /* 9F DAI2 EQ3 */ + + { 0xa0, 0x00 }, /* A0 DAI2 EQ3 */ + { 0xa1, 0x00 }, /* A1 DAI2 EQ3 */ + { 0xa2, 0x00 }, /* A2 DAI2 EQ4 */ + { 0xa3, 0x00 }, /* A3 DAI2 EQ4 */ + { 0xa4, 0x00 }, /* A4 DAI2 EQ4 */ + { 0xa5, 0x00 }, /* A5 DAI2 EQ4 */ + { 0xa6, 0x00 }, /* A6 DAI2 EQ4 */ + { 0xa7, 0x00 }, /* A7 DAI2 EQ4 */ + { 0xa8, 0x00 }, /* A8 DAI2 EQ4 */ + { 0xa9, 0x00 }, /* A9 DAI2 EQ4 */ + { 0xaa, 0x00 }, /* AA DAI2 EQ4 */ + { 0xab, 0x00 }, /* AB DAI2 EQ4 */ + { 0xac, 0x00 }, /* AC DAI2 EQ5 */ + { 0xad, 0x00 }, /* AD DAI2 EQ5 */ + { 0xae, 0x00 }, /* AE DAI2 EQ5 */ + { 0xaf, 0x00 }, /* AF DAI2 EQ5 */ + + { 0xb0, 0x00 }, /* B0 DAI2 EQ5 */ + { 0xb1, 0x00 }, /* B1 DAI2 EQ5 */ + { 0xb2, 0x00 }, /* B2 DAI2 EQ5 */ + { 0xb3, 0x00 }, /* B3 DAI2 EQ5 */ + { 0xb4, 0x00 }, /* B4 DAI2 EQ5 */ + { 0xb5, 0x00 }, /* B5 DAI2 EQ5 */ + { 0xb6, 0x00 }, /* B6 DAI1 biquad */ + { 0xb7, 0x00 }, /* B7 DAI1 biquad */ + { 0xb8 ,0x00 }, /* B8 DAI1 biquad */ + { 0xb9, 0x00 }, /* B9 DAI1 biquad */ + { 0xba, 0x00 }, /* BA DAI1 biquad */ + { 0xbb, 0x00 }, /* BB DAI1 biquad */ + { 0xbc, 0x00 }, /* BC DAI1 biquad */ + { 0xbd, 0x00 }, /* BD DAI1 biquad */ + { 0xbe, 0x00 }, /* BE DAI1 biquad */ + { 0xbf, 0x00 }, /* BF DAI1 biquad */ + + { 0xc0, 0x00 }, /* C0 DAI2 biquad */ + { 0xc1, 0x00 }, /* C1 DAI2 biquad */ + { 0xc2, 0x00 }, /* C2 DAI2 biquad */ + { 0xc3, 0x00 }, /* C3 DAI2 biquad */ + { 0xc4, 0x00 }, /* C4 DAI2 biquad */ + { 0xc5, 0x00 }, /* C5 DAI2 biquad */ + { 0xc6, 0x00 }, /* C6 DAI2 biquad */ + { 0xc7, 0x00 }, /* C7 DAI2 biquad */ + { 0xc8, 0x00 }, /* C8 DAI2 biquad */ + { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; static struct { @@ -606,11 +536,27 @@ static struct { { 0xFF, 0x00, 1 }, /* FF */ }; -static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool max98088_readable_register(struct device *dev, unsigned int reg) +{ + return max98088_access[reg].readable; +} + +static bool max98088_volatile_register(struct device *dev, unsigned int reg) { return max98088_access[reg].vol; } +static const struct regmap_config max98088_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max98088_readable_register, + .volatile_reg = max98088_volatile_register, + + .reg_defaults = max98088_reg, + .num_reg_defaults = ARRAY_SIZE(max98088_reg), + .cache_type = REGCACHE_RBTREE, +}; /* * Load equalizer DSP coefficient configurations registers @@ -1610,58 +1556,34 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static void max98088_sync_cache(struct snd_soc_codec *codec) -{ - u8 *reg_cache = codec->reg_cache; - int i; - - if (!codec->cache_sync) - return; - - codec->cache_only = 0; - - /* write back cached values if they're writeable and - * different from the hardware default. - */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - if (!max98088_access[i].writable) - continue; - - if (reg_cache[i] == max98088_reg[i]) - continue; - - snd_soc_write(codec, i, reg_cache[i]); - } - - codec->cache_sync = 0; -} - static int max98088_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - switch (level) { - case SND_SOC_BIAS_ON: - break; - - case SND_SOC_BIAS_PREPARE: - break; - - case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - max98088_sync_cache(codec); - - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, - M98088_MBEN, M98088_MBEN); - break; - - case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, - M98088_MBEN, 0); - codec->cache_sync = 1; - break; - } - codec->dapm.bias_level = level; - return 0; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(max98088->regmap); + + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, M98088_MBEN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, 0); + regcache_mark_dirty(max98088->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; } #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000 @@ -1988,9 +1910,9 @@ static int max98088_probe(struct snd_soc_codec *codec) struct max98088_cdata *cdata; int ret = 0; - codec->cache_sync = 1; + regcache_mark_dirty(max98088->regmap); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -2070,10 +1992,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { .set_bias_level = max98088_set_bias_level, .controls = max98088_snd_controls, .num_controls = ARRAY_SIZE(max98088_snd_controls), - .reg_cache_size = ARRAY_SIZE(max98088_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98088_reg, - .volatile_register = max98088_volatile_register, .dapm_widgets = max98088_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), .dapm_routes = max98088_audio_map, @@ -2081,7 +1999,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { }; static int max98088_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *id) { struct max98088_priv *max98088; int ret; @@ -2091,6 +2009,10 @@ static int max98088_i2c_probe(struct i2c_client *i2c, if (max98088 == NULL) return -ENOMEM; + max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap); + if (IS_ERR(max98088->regmap)) + return PTR_ERR(max98088->regmap); + max98088->devtype = id->driver_data; i2c_set_clientdata(i2c, max98088); -- cgit From 2245e3c31c15c2d2a26926c4b734f4d3a37ae252 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 11:50:10 +0100 Subject: ASoC: ab8500: Explicitly set I/O up We do some I/O in probe so we need to ensure the I/O operations are fully set up then. Reported-by: Olof Johansson Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index d5a0fc4b2fe2..7f6ca111659b 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2468,6 +2468,8 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); -- cgit From d36126ac5674a83e1d426877709437b24f058f47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 18:58:59 +0100 Subject: ASoC: max98095: Remove custom hw_write() implementation The registers that are being kept uncached are marked as volatile anyway so the call has no practical impact. Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 25 ++++--------------------- 1 file changed, 4 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd1642970..65aba5ec52df 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -611,23 +611,6 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) return 0; } -/* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; - - codec->cache_bypass = 1; - ret = snd_soc_write(codec, reg, value); - codec->cache_bypass = 0; - - return ret ? -EIO : 0; -} - /* * Load equalizer DSP coefficient configurations registers */ @@ -648,8 +631,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -673,8 +656,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } -- cgit From c6b3283f6d8818177349eb7cc0549286a55140c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:05:16 +0100 Subject: ASoC: max90895: Convert to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 65aba5ec52df..1a4585ae36e4 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1268,14 +1268,6 @@ static const struct snd_soc_dapm_route max98095_audio_map[] = { {"MIC2 Input", NULL, "MIC2"}, }; -static int max98095_add_widgets(struct snd_soc_codec *codec) -{ - snd_soc_add_codec_controls(codec, max98095_snd_controls, - ARRAY_SIZE(max98095_snd_controls)); - - return 0; -} - /* codec mclk clock divider coefficients */ static const struct { u32 rate; @@ -2430,8 +2422,6 @@ static int max98095_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, M98095_097_PWR_SYS, M98095_SHDNRUN, M98095_SHDNRUN); - max98095_add_widgets(codec); - return 0; err_irq: @@ -2463,6 +2453,8 @@ static struct snd_soc_codec_driver soc_codec_dev_max98095 = { .suspend = max98095_suspend, .resume = max98095_resume, .set_bias_level = max98095_set_bias_level, + .controls = max98095_snd_controls, + .num_controls = ARRAY_SIZE(max98095_snd_controls), .reg_cache_size = ARRAY_SIZE(max98095_reg_def), .reg_word_size = sizeof(u8), .reg_cache_default = max98095_reg_def, -- cgit From 14acbbbbc649c4c6057f601396b8000cd616d9ad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:08:35 +0100 Subject: ASoC: max98095: Convert to direct regmap API usage Saves code and moves us towards being able to remove the duplicate ASoC level register I/O functionality. Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 435 +++++++++++++++++--------------------------- 1 file changed, 167 insertions(+), 268 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 1a4585ae36e4..5c9f6b527cf0 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -39,6 +39,7 @@ struct max98095_cdata { }; struct max98095_priv { + struct regmap *regmap; enum max98095_type devtype; struct max98095_pdata *pdata; unsigned int sysclk; @@ -56,263 +57,145 @@ struct max98095_priv { struct snd_soc_jack *mic_jack; }; -static const u8 max98095_reg_def[M98095_REG_CNT] = { - 0x00, /* 00 */ - 0x00, /* 01 */ - 0x00, /* 02 */ - 0x00, /* 03 */ - 0x00, /* 04 */ - 0x00, /* 05 */ - 0x00, /* 06 */ - 0x00, /* 07 */ - 0x00, /* 08 */ - 0x00, /* 09 */ - 0x00, /* 0A */ - 0x00, /* 0B */ - 0x00, /* 0C */ - 0x00, /* 0D */ - 0x00, /* 0E */ - 0x00, /* 0F */ - 0x00, /* 10 */ - 0x00, /* 11 */ - 0x00, /* 12 */ - 0x00, /* 13 */ - 0x00, /* 14 */ - 0x00, /* 15 */ - 0x00, /* 16 */ - 0x00, /* 17 */ - 0x00, /* 18 */ - 0x00, /* 19 */ - 0x00, /* 1A */ - 0x00, /* 1B */ - 0x00, /* 1C */ - 0x00, /* 1D */ - 0x00, /* 1E */ - 0x00, /* 1F */ - 0x00, /* 20 */ - 0x00, /* 21 */ - 0x00, /* 22 */ - 0x00, /* 23 */ - 0x00, /* 24 */ - 0x00, /* 25 */ - 0x00, /* 26 */ - 0x00, /* 27 */ - 0x00, /* 28 */ - 0x00, /* 29 */ - 0x00, /* 2A */ - 0x00, /* 2B */ - 0x00, /* 2C */ - 0x00, /* 2D */ - 0x00, /* 2E */ - 0x00, /* 2F */ - 0x00, /* 30 */ - 0x00, /* 31 */ - 0x00, /* 32 */ - 0x00, /* 33 */ - 0x00, /* 34 */ - 0x00, /* 35 */ - 0x00, /* 36 */ - 0x00, /* 37 */ - 0x00, /* 38 */ - 0x00, /* 39 */ - 0x00, /* 3A */ - 0x00, /* 3B */ - 0x00, /* 3C */ - 0x00, /* 3D */ - 0x00, /* 3E */ - 0x00, /* 3F */ - 0x00, /* 40 */ - 0x00, /* 41 */ - 0x00, /* 42 */ - 0x00, /* 43 */ - 0x00, /* 44 */ - 0x00, /* 45 */ - 0x00, /* 46 */ - 0x00, /* 47 */ - 0x00, /* 48 */ - 0x00, /* 49 */ - 0x00, /* 4A */ - 0x00, /* 4B */ - 0x00, /* 4C */ - 0x00, /* 4D */ - 0x00, /* 4E */ - 0x00, /* 4F */ - 0x00, /* 50 */ - 0x00, /* 51 */ - 0x00, /* 52 */ - 0x00, /* 53 */ - 0x00, /* 54 */ - 0x00, /* 55 */ - 0x00, /* 56 */ - 0x00, /* 57 */ - 0x00, /* 58 */ - 0x00, /* 59 */ - 0x00, /* 5A */ - 0x00, /* 5B */ - 0x00, /* 5C */ - 0x00, /* 5D */ - 0x00, /* 5E */ - 0x00, /* 5F */ - 0x00, /* 60 */ - 0x00, /* 61 */ - 0x00, /* 62 */ - 0x00, /* 63 */ - 0x00, /* 64 */ - 0x00, /* 65 */ - 0x00, /* 66 */ - 0x00, /* 67 */ - 0x00, /* 68 */ - 0x00, /* 69 */ - 0x00, /* 6A */ - 0x00, /* 6B */ - 0x00, /* 6C */ - 0x00, /* 6D */ - 0x00, /* 6E */ - 0x00, /* 6F */ - 0x00, /* 70 */ - 0x00, /* 71 */ - 0x00, /* 72 */ - 0x00, /* 73 */ - 0x00, /* 74 */ - 0x00, /* 75 */ - 0x00, /* 76 */ - 0x00, /* 77 */ - 0x00, /* 78 */ - 0x00, /* 79 */ - 0x00, /* 7A */ - 0x00, /* 7B */ - 0x00, /* 7C */ - 0x00, /* 7D */ - 0x00, /* 7E */ - 0x00, /* 7F */ - 0x00, /* 80 */ - 0x00, /* 81 */ - 0x00, /* 82 */ - 0x00, /* 83 */ - 0x00, /* 84 */ - 0x00, /* 85 */ - 0x00, /* 86 */ - 0x00, /* 87 */ - 0x00, /* 88 */ - 0x00, /* 89 */ - 0x00, /* 8A */ - 0x00, /* 8B */ - 0x00, /* 8C */ - 0x00, /* 8D */ - 0x00, /* 8E */ - 0x00, /* 8F */ - 0x00, /* 90 */ - 0x00, /* 91 */ - 0x30, /* 92 */ - 0xF0, /* 93 */ - 0x00, /* 94 */ - 0x00, /* 95 */ - 0x3F, /* 96 */ - 0x00, /* 97 */ - 0x00, /* 98 */ - 0x00, /* 99 */ - 0x00, /* 9A */ - 0x00, /* 9B */ - 0x00, /* 9C */ - 0x00, /* 9D */ - 0x00, /* 9E */ - 0x00, /* 9F */ - 0x00, /* A0 */ - 0x00, /* A1 */ - 0x00, /* A2 */ - 0x00, /* A3 */ - 0x00, /* A4 */ - 0x00, /* A5 */ - 0x00, /* A6 */ - 0x00, /* A7 */ - 0x00, /* A8 */ - 0x00, /* A9 */ - 0x00, /* AA */ - 0x00, /* AB */ - 0x00, /* AC */ - 0x00, /* AD */ - 0x00, /* AE */ - 0x00, /* AF */ - 0x00, /* B0 */ - 0x00, /* B1 */ - 0x00, /* B2 */ - 0x00, /* B3 */ - 0x00, /* B4 */ - 0x00, /* B5 */ - 0x00, /* B6 */ - 0x00, /* B7 */ - 0x00, /* B8 */ - 0x00, /* B9 */ - 0x00, /* BA */ - 0x00, /* BB */ - 0x00, /* BC */ - 0x00, /* BD */ - 0x00, /* BE */ - 0x00, /* BF */ - 0x00, /* C0 */ - 0x00, /* C1 */ - 0x00, /* C2 */ - 0x00, /* C3 */ - 0x00, /* C4 */ - 0x00, /* C5 */ - 0x00, /* C6 */ - 0x00, /* C7 */ - 0x00, /* C8 */ - 0x00, /* C9 */ - 0x00, /* CA */ - 0x00, /* CB */ - 0x00, /* CC */ - 0x00, /* CD */ - 0x00, /* CE */ - 0x00, /* CF */ - 0x00, /* D0 */ - 0x00, /* D1 */ - 0x00, /* D2 */ - 0x00, /* D3 */ - 0x00, /* D4 */ - 0x00, /* D5 */ - 0x00, /* D6 */ - 0x00, /* D7 */ - 0x00, /* D8 */ - 0x00, /* D9 */ - 0x00, /* DA */ - 0x00, /* DB */ - 0x00, /* DC */ - 0x00, /* DD */ - 0x00, /* DE */ - 0x00, /* DF */ - 0x00, /* E0 */ - 0x00, /* E1 */ - 0x00, /* E2 */ - 0x00, /* E3 */ - 0x00, /* E4 */ - 0x00, /* E5 */ - 0x00, /* E6 */ - 0x00, /* E7 */ - 0x00, /* E8 */ - 0x00, /* E9 */ - 0x00, /* EA */ - 0x00, /* EB */ - 0x00, /* EC */ - 0x00, /* ED */ - 0x00, /* EE */ - 0x00, /* EF */ - 0x00, /* F0 */ - 0x00, /* F1 */ - 0x00, /* F2 */ - 0x00, /* F3 */ - 0x00, /* F4 */ - 0x00, /* F5 */ - 0x00, /* F6 */ - 0x00, /* F7 */ - 0x00, /* F8 */ - 0x00, /* F9 */ - 0x00, /* FA */ - 0x00, /* FB */ - 0x00, /* FC */ - 0x00, /* FD */ - 0x00, /* FE */ - 0x00, /* FF */ +static const struct reg_default max98095_reg_def[] = { + { 0xf, 0x00 }, /* 0F */ + { 0x10, 0x00 }, /* 10 */ + { 0x11, 0x00 }, /* 11 */ + { 0x12, 0x00 }, /* 12 */ + { 0x13, 0x00 }, /* 13 */ + { 0x14, 0x00 }, /* 14 */ + { 0x15, 0x00 }, /* 15 */ + { 0x16, 0x00 }, /* 16 */ + { 0x17, 0x00 }, /* 17 */ + { 0x18, 0x00 }, /* 18 */ + { 0x19, 0x00 }, /* 19 */ + { 0x1a, 0x00 }, /* 1A */ + { 0x1b, 0x00 }, /* 1B */ + { 0x1c, 0x00 }, /* 1C */ + { 0x1d, 0x00 }, /* 1D */ + { 0x1e, 0x00 }, /* 1E */ + { 0x1f, 0x00 }, /* 1F */ + { 0x20, 0x00 }, /* 20 */ + { 0x21, 0x00 }, /* 21 */ + { 0x22, 0x00 }, /* 22 */ + { 0x23, 0x00 }, /* 23 */ + { 0x24, 0x00 }, /* 24 */ + { 0x25, 0x00 }, /* 25 */ + { 0x26, 0x00 }, /* 26 */ + { 0x27, 0x00 }, /* 27 */ + { 0x28, 0x00 }, /* 28 */ + { 0x29, 0x00 }, /* 29 */ + { 0x2a, 0x00 }, /* 2A */ + { 0x2b, 0x00 }, /* 2B */ + { 0x2c, 0x00 }, /* 2C */ + { 0x2d, 0x00 }, /* 2D */ + { 0x2e, 0x00 }, /* 2E */ + { 0x2f, 0x00 }, /* 2F */ + { 0x30, 0x00 }, /* 30 */ + { 0x31, 0x00 }, /* 31 */ + { 0x32, 0x00 }, /* 32 */ + { 0x33, 0x00 }, /* 33 */ + { 0x34, 0x00 }, /* 34 */ + { 0x35, 0x00 }, /* 35 */ + { 0x36, 0x00 }, /* 36 */ + { 0x37, 0x00 }, /* 37 */ + { 0x38, 0x00 }, /* 38 */ + { 0x39, 0x00 }, /* 39 */ + { 0x3a, 0x00 }, /* 3A */ + { 0x3b, 0x00 }, /* 3B */ + { 0x3c, 0x00 }, /* 3C */ + { 0x3d, 0x00 }, /* 3D */ + { 0x3e, 0x00 }, /* 3E */ + { 0x3f, 0x00 }, /* 3F */ + { 0x40, 0x00 }, /* 40 */ + { 0x41, 0x00 }, /* 41 */ + { 0x42, 0x00 }, /* 42 */ + { 0x43, 0x00 }, /* 43 */ + { 0x44, 0x00 }, /* 44 */ + { 0x45, 0x00 }, /* 45 */ + { 0x46, 0x00 }, /* 46 */ + { 0x47, 0x00 }, /* 47 */ + { 0x48, 0x00 }, /* 48 */ + { 0x49, 0x00 }, /* 49 */ + { 0x4a, 0x00 }, /* 4A */ + { 0x4b, 0x00 }, /* 4B */ + { 0x4c, 0x00 }, /* 4C */ + { 0x4d, 0x00 }, /* 4D */ + { 0x4e, 0x00 }, /* 4E */ + { 0x4f, 0x00 }, /* 4F */ + { 0x50, 0x00 }, /* 50 */ + { 0x51, 0x00 }, /* 51 */ + { 0x52, 0x00 }, /* 52 */ + { 0x53, 0x00 }, /* 53 */ + { 0x54, 0x00 }, /* 54 */ + { 0x55, 0x00 }, /* 55 */ + { 0x56, 0x00 }, /* 56 */ + { 0x57, 0x00 }, /* 57 */ + { 0x58, 0x00 }, /* 58 */ + { 0x59, 0x00 }, /* 59 */ + { 0x5a, 0x00 }, /* 5A */ + { 0x5b, 0x00 }, /* 5B */ + { 0x5c, 0x00 }, /* 5C */ + { 0x5d, 0x00 }, /* 5D */ + { 0x5e, 0x00 }, /* 5E */ + { 0x5f, 0x00 }, /* 5F */ + { 0x60, 0x00 }, /* 60 */ + { 0x61, 0x00 }, /* 61 */ + { 0x62, 0x00 }, /* 62 */ + { 0x63, 0x00 }, /* 63 */ + { 0x64, 0x00 }, /* 64 */ + { 0x65, 0x00 }, /* 65 */ + { 0x66, 0x00 }, /* 66 */ + { 0x67, 0x00 }, /* 67 */ + { 0x68, 0x00 }, /* 68 */ + { 0x69, 0x00 }, /* 69 */ + { 0x6a, 0x00 }, /* 6A */ + { 0x6b, 0x00 }, /* 6B */ + { 0x6c, 0x00 }, /* 6C */ + { 0x6d, 0x00 }, /* 6D */ + { 0x6e, 0x00 }, /* 6E */ + { 0x6f, 0x00 }, /* 6F */ + { 0x70, 0x00 }, /* 70 */ + { 0x71, 0x00 }, /* 71 */ + { 0x72, 0x00 }, /* 72 */ + { 0x73, 0x00 }, /* 73 */ + { 0x74, 0x00 }, /* 74 */ + { 0x75, 0x00 }, /* 75 */ + { 0x76, 0x00 }, /* 76 */ + { 0x77, 0x00 }, /* 77 */ + { 0x78, 0x00 }, /* 78 */ + { 0x79, 0x00 }, /* 79 */ + { 0x7a, 0x00 }, /* 7A */ + { 0x7b, 0x00 }, /* 7B */ + { 0x7c, 0x00 }, /* 7C */ + { 0x7d, 0x00 }, /* 7D */ + { 0x7e, 0x00 }, /* 7E */ + { 0x7f, 0x00 }, /* 7F */ + { 0x80, 0x00 }, /* 80 */ + { 0x81, 0x00 }, /* 81 */ + { 0x82, 0x00 }, /* 82 */ + { 0x83, 0x00 }, /* 83 */ + { 0x84, 0x00 }, /* 84 */ + { 0x85, 0x00 }, /* 85 */ + { 0x86, 0x00 }, /* 86 */ + { 0x87, 0x00 }, /* 87 */ + { 0x88, 0x00 }, /* 88 */ + { 0x89, 0x00 }, /* 89 */ + { 0x8a, 0x00 }, /* 8A */ + { 0x8b, 0x00 }, /* 8B */ + { 0x8c, 0x00 }, /* 8C */ + { 0x8d, 0x00 }, /* 8D */ + { 0x8e, 0x00 }, /* 8E */ + { 0x8f, 0x00 }, /* 8F */ + { 0x90, 0x00 }, /* 90 */ + { 0x91, 0x00 }, /* 91 */ + { 0x92, 0x30 }, /* 92 */ + { 0x93, 0xF0 }, /* 93 */ + { 0x94, 0x00 }, /* 94 */ + { 0x95, 0x00 }, /* 95 */ + { 0x96, 0x3F }, /* 96 */ + { 0x97, 0x00 }, /* 97 */ + { 0xff, 0x00 }, /* FF */ }; static struct { @@ -577,14 +460,14 @@ static struct { { 0xFF, 0x00 }, /* FF */ }; -static int max98095_readable(struct snd_soc_codec *codec, unsigned int reg) +static bool max98095_readable(struct device *dev, unsigned int reg) { if (reg >= M98095_REG_CNT) return 0; return max98095_access[reg].readable != 0; } -static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool max98095_volatile(struct device *dev, unsigned int reg) { if (reg > M98095_REG_MAX_CACHED) return 1; @@ -611,6 +494,19 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) return 0; } +static const struct regmap_config max98095_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max98095_reg_def, + .num_reg_defaults = ARRAY_SIZE(max98095_reg_def), + .max_register = M98095_0FF_REV_ID, + .cache_type = REGCACHE_RBTREE, + + .readable_reg = max98095_readable, + .volatile_reg = max98095_volatile, +}; + /* * Load equalizer DSP coefficient configurations registers */ @@ -1723,6 +1619,7 @@ static int max98095_dai3_set_fmt(struct snd_soc_dai *codec_dai, static int max98095_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -1734,7 +1631,7 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max98095->regmap); if (ret != 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -1749,7 +1646,7 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, M98095_090_PWR_EN_IN, M98095_MBEN, 0); - codec->cache_sync = 1; + regcache_mark_dirty(max98095->regmap); break; } codec->dapm.bias_level = level; @@ -2316,7 +2213,7 @@ static int max98095_reset(struct snd_soc_codec *codec) /* Reset to hardware default for registers, as there is not * a soft reset hardware control register */ for (i = M98095_010_HOST_INT_CFG; i < M98095_REG_MAX_CACHED; i++) { - ret = snd_soc_write(codec, i, max98095_reg_def[i]); + ret = snd_soc_write(codec, i, snd_soc_read(codec, i)); if (ret < 0) { dev_err(codec->dev, "Failed to reset: %d\n", ret); return ret; @@ -2333,7 +2230,7 @@ static int max98095_probe(struct snd_soc_codec *codec) struct i2c_client *client; int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -2455,11 +2352,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max98095 = { .set_bias_level = max98095_set_bias_level, .controls = max98095_snd_controls, .num_controls = ARRAY_SIZE(max98095_snd_controls), - .reg_cache_size = ARRAY_SIZE(max98095_reg_def), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98095_reg_def, - .readable_register = max98095_readable, - .volatile_register = max98095_volatile, .dapm_widgets = max98095_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98095_dapm_widgets), .dapm_routes = max98095_audio_map, @@ -2477,6 +2369,13 @@ static int max98095_i2c_probe(struct i2c_client *i2c, if (max98095 == NULL) return -ENOMEM; + max98095->regmap = devm_regmap_init_i2c(i2c, &max98095_regmap); + if (IS_ERR(max98095->regmap)) { + ret = PTR_ERR(max98095->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + max98095->devtype = id->driver_data; i2c_set_clientdata(i2c, max98095); max98095->pdata = i2c->dev.platform_data; -- cgit From 2c142c61f79c14a120c0f4d2954e35b6404b2d0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 18:49:54 +0100 Subject: ASoC: tlv320aic23: Remove #defines for I2C The only control interface supported by this driver is I2C so there is no need for conditional compilation around the control interface. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 31762ebdd774..32994597a43f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -613,7 +613,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -660,29 +659,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .id_table = tlv320aic23_id, }; -#endif - -static int __init tlv320aic23_modinit(void) -{ - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&tlv320aic23_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(tlv320aic23_modinit); - -static void __exit tlv320aic23_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&tlv320aic23_i2c_driver); -#endif -} -module_exit(tlv320aic23_exit); +module_i2c_driver(tlv320aic23_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS "); -- cgit From b07c443fabb97f909c8cc406bfd2d0ecc002bc3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 18:51:26 +0100 Subject: ASoC: tlv320aic23: Convert to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 32994597a43f..3a6be8c3d557 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -586,9 +586,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); - snd_soc_add_codec_controls(codec, tlv320aic23_snd_controls, - ARRAY_SIZE(tlv320aic23_snd_controls)); - return 0; } @@ -607,6 +604,8 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, .set_bias_level = tlv320aic23_set_bias_level, + .controls = tlv320aic23_snd_controls, + .num_controls = ARRAY_SIZE(tlv320aic23_snd_controls), .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), .dapm_routes = tlv320aic23_intercon, -- cgit From a16bbe4d685c1465b98d3fabdb95310eafcd383e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 20:07:12 +0100 Subject: ASoC: tlv320aic3x: Remove nonsense comment for register cache Every statement in this comment is incorrect either through bitrot or (mostly) through never having corresponded to reality in the first place. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6e3f269243e0..3abbff3fe888 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -90,12 +90,6 @@ struct aic3x_priv { enum aic3x_micbias_voltage micbias_vg; }; -/* - * AIC3X register cache - * We can't read the AIC3X register space when we are - * using 2 wire for device control, so we cache them instead. - * There is no point in caching the reset register - */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x10, /* 0 */ 0x04, 0x00, 0x00, 0x00, /* 4 */ -- cgit From 6f818e04fc8d3d413eeb3a689c7607f2a89ab568 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:48:45 +0100 Subject: ASoC: tlv320aic3x: Move resource acquisition to I2C probe This is more idiomatic and interacts better with deferred probing. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 63 ++++++++++++++++++++++-------------------- 1 file changed, 33 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3abbff3fe888..de17a36beb6f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1345,23 +1345,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) return ret; } - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) { - ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset"); - if (ret != 0) - goto err_gpio; - gpio_direction_output(aic3x->gpio_reset, 0); - } - - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) - aic3x->supplies[i].supply = aic3x_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3x->supplies), - aic3x->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err_get; - } for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; @@ -1418,12 +1401,6 @@ err_notif: while (i--) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); - regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); -err_get: - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) - gpio_free(aic3x->gpio_reset); -err_gpio: return ret; } @@ -1434,15 +1411,9 @@ static int aic3x_remove(struct snd_soc_codec *codec) aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); list_del(&aic3x->list); - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) { - gpio_set_value(aic3x->gpio_reset, 0); - gpio_free(aic3x->gpio_reset); - } for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); - regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); return 0; } @@ -1484,7 +1455,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_priv *aic3x; struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; - int ret; + int ret, i; u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1545,14 +1516,46 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->model = id->driver_data; + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err; + gpio_direction_output(aic3x->gpio_reset, 0); + } + + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) + aic3x->supplies[i].supply = aic3x_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(aic3x->supplies), + aic3x->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + goto err_gpio; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); return ret; + +err_gpio: + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) + gpio_free(aic3x->gpio_reset); +err: + return ret; } static int aic3x_i2c_remove(struct i2c_client *client) { + struct aic3x_priv *aic3x = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } return 0; } -- cgit From f9df1ae6b59e5bb16d3094e9c1c8b6feeaf32aae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 23:53:16 +0100 Subject: ASoC: tlv320aic3x: Move to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index de17a36beb6f..397a2133e2d1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1369,8 +1369,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) (aic3x->setup->gpio_func[1] & 0xf) << 4); } - snd_soc_add_codec_controls(codec, aic3x_snd_controls, - ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); @@ -1428,6 +1426,8 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .remove = aic3x_remove, .suspend = aic3x_suspend, .resume = aic3x_resume, + .controls = aic3x_snd_controls, + .num_controls = ARRAY_SIZE(aic3x_snd_controls), }; /* -- cgit From 58a63fbd7c80510140a94442b2ca9199bb6d51c3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 23:57:36 +0100 Subject: ASoC: tlv320aic3x: Move to table based DAPM init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 397a2133e2d1..16fc74cae754 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -818,12 +818,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); - - /* set up audio path interconnects */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - if (aic3x->model == AIC3X_MODEL_3007) { snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); @@ -1428,6 +1422,10 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .resume = aic3x_resume, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), + .dapm_widgets = aic3x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; /* -- cgit From 2677b4bb7316c07dd53535e01bd9b2ec699d0314 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:55:39 +0100 Subject: ASoC: tlv320aic3x: Don't reference cache datastructure directly Rather than referencing the cache directly read back the values we are going to restore, supporting refactoring to use regmap. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 16fc74cae754..83e7d855c49a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1068,14 +1068,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int aic3x_init_3007(struct snd_soc_codec *codec) { - u8 tmp1, tmp2, *cache = codec->reg_cache; + unsigned int tmp1, tmp2; /* * There is no need to cache writes to undocumented page 0xD but * respective page 0 register cache entries must be preserved */ - tmp1 = cache[0xD]; - tmp2 = cache[0x8]; + tmp1 = snd_soc_read(codec, 0xD); + tmp2 = snd_soc_read(codec, 0x8); /* Class-D speaker driver init; datasheet p. 46 */ snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); snd_soc_write(codec, 0xD, 0x0D); @@ -1083,8 +1083,9 @@ static int aic3x_init_3007(struct snd_soc_codec *codec) snd_soc_write(codec, 0x8, 0x5D); snd_soc_write(codec, 0x8, 0x5C); snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); - cache[0xD] = tmp1; - cache[0x8] = tmp2; + + snd_soc_write(codec, 0xD, tmp1); + snd_soc_write(codec, 0x8, tmp2); return 0; } -- cgit From 2a6fedec195b9bd20e60f9825ba7cc6315e54652 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 00:07:13 +0100 Subject: ASoC: tlv320aic3x: Convert to direct regmap API usage This is slightly more complex than a standard regmap conversion due to the moderately detailed cache control and the open coding of a register patch for the class D speaker on the TLV320AIC3007. Although the device supports paging this is not currently implemented as the additional pages are only used during the application of the patch for the TLV320AIC3007. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 151 ++++++++++++++++++++--------------------- 1 file changed, 73 insertions(+), 78 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 83e7d855c49a..892c108ca67a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -72,9 +72,9 @@ struct aic3x_disable_nb { /* codec private data */ struct aic3x_priv { struct snd_soc_codec *codec; + struct regmap *regmap; struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; - enum snd_soc_control_type control_type; struct aic3x_setup_data *setup; unsigned int sysclk; struct list_head list; @@ -90,35 +90,45 @@ struct aic3x_priv { enum aic3x_micbias_voltage micbias_vg; }; -static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { - 0x00, 0x00, 0x00, 0x10, /* 0 */ - 0x04, 0x00, 0x00, 0x00, /* 4 */ - 0x00, 0x00, 0x00, 0x01, /* 8 */ - 0x00, 0x00, 0x00, 0x80, /* 12 */ - 0x80, 0xff, 0xff, 0x78, /* 16 */ - 0x78, 0x78, 0x78, 0x78, /* 20 */ - 0x78, 0x00, 0x00, 0xfe, /* 24 */ - 0x00, 0x00, 0xfe, 0x00, /* 28 */ - 0x18, 0x18, 0x00, 0x00, /* 32 */ - 0x00, 0x00, 0x00, 0x00, /* 36 */ - 0x00, 0x00, 0x00, 0x80, /* 40 */ - 0x80, 0x00, 0x00, 0x00, /* 44 */ - 0x00, 0x00, 0x00, 0x04, /* 48 */ - 0x00, 0x00, 0x00, 0x00, /* 52 */ - 0x00, 0x00, 0x04, 0x00, /* 56 */ - 0x00, 0x00, 0x00, 0x00, /* 60 */ - 0x00, 0x04, 0x00, 0x00, /* 64 */ - 0x00, 0x00, 0x00, 0x00, /* 68 */ - 0x04, 0x00, 0x00, 0x00, /* 72 */ - 0x00, 0x00, 0x00, 0x00, /* 76 */ - 0x00, 0x00, 0x00, 0x00, /* 80 */ - 0x00, 0x00, 0x00, 0x00, /* 84 */ - 0x00, 0x00, 0x00, 0x00, /* 88 */ - 0x00, 0x00, 0x00, 0x00, /* 92 */ - 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, 0x00, /* 100 */ - 0x00, 0x00, 0x00, 0x00, /* 104 */ - 0x00, 0x00, /* 108 */ +static const struct reg_default aic3x_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x10 }, + { 4, 0x04 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x01 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x80 }, + { 16, 0x80 }, { 17, 0xff }, { 18, 0xff }, { 19, 0x78 }, + { 20, 0x78 }, { 21, 0x78 }, { 22, 0x78 }, { 23, 0x78 }, + { 24, 0x78 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0xfe }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0xfe }, { 31, 0x00 }, + { 32, 0x18 }, { 33, 0x18 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x00 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x80 }, + { 44, 0x80 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x04 }, + { 52, 0x00 }, { 53, 0x00 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x04 }, { 59, 0x00 }, + { 60, 0x00 }, { 61, 0x00 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x04 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x04 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x00 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x00 }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x00 }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x02 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, +}; + +static const struct regmap_config aic3x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DAC_ICC_ADJ, + .reg_defaults = aic3x_reg, + .num_reg_defaults = ARRAY_SIZE(aic3x_reg), + .cache_type = REGCACHE_RBTREE, }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ @@ -1066,30 +1076,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int aic3x_init_3007(struct snd_soc_codec *codec) -{ - unsigned int tmp1, tmp2; - - /* - * There is no need to cache writes to undocumented page 0xD but - * respective page 0 register cache entries must be preserved - */ - tmp1 = snd_soc_read(codec, 0xD); - tmp2 = snd_soc_read(codec, 0x8); - /* Class-D speaker driver init; datasheet p. 46 */ - snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); - snd_soc_write(codec, 0xD, 0x0D); - snd_soc_write(codec, 0x8, 0x5C); - snd_soc_write(codec, 0x8, 0x5D); - snd_soc_write(codec, 0x8, 0x5C); - snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); - - snd_soc_write(codec, 0xD, tmp1); - snd_soc_write(codec, 0x8, tmp2); - - return 0; -} - static int aic3x_regulator_event(struct notifier_block *nb, unsigned long event, void *data) { @@ -1104,7 +1090,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, */ if (gpio_is_valid(aic3x->gpio_reset)) gpio_set_value(aic3x->gpio_reset, 0); - aic3x->codec->cache_sync = 1; + regcache_mark_dirty(aic3x->regmap); } return 0; @@ -1113,8 +1099,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, static int aic3x_set_power(struct snd_soc_codec *codec, int power) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - int i, ret; - u8 *cache = codec->reg_cache; + int ret; if (power) { ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies), @@ -1122,12 +1107,6 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) if (ret) goto out; aic3x->power = 1; - /* - * Reset release and cache sync is necessary only if some - * supply was off or if there were cached writes - */ - if (!codec->cache_sync) - goto out; if (gpio_is_valid(aic3x->gpio_reset)) { udelay(1); @@ -1135,12 +1114,8 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) } /* Sync reg_cache with the hardware */ - codec->cache_only = 0; - for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) - snd_soc_write(codec, i, cache[i]); - if (aic3x->model == AIC3X_MODEL_3007) - aic3x_init_3007(codec); - codec->cache_sync = 0; + regcache_cache_only(aic3x->regmap, false); + regcache_sync(aic3x->regmap); } else { /* * Do soft reset to this codec instance in order to clear @@ -1148,10 +1123,10 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) * remain on */ snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); - codec->cache_sync = 1; + regcache_mark_dirty(aic3x->regmap); aic3x->power = 0; /* HW writes are needless when bias is off */ - codec->cache_only = 1; + regcache_cache_only(aic3x->regmap, true); ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); } @@ -1306,7 +1281,6 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { - aic3x_init_3007(codec); snd_soc_write(codec, CLASSD_CTRL, 0); } @@ -1334,7 +1308,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1353,7 +1327,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) } } - codec->cache_only = 1; + regcache_mark_dirty(aic3x->regmap); aic3x_init(codec); if (aic3x->setup) { @@ -1414,9 +1388,6 @@ static int aic3x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(aic3x_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = aic3x_reg, .probe = aic3x_probe, .remove = aic3x_remove, .suspend = aic3x_suspend, @@ -1443,6 +1414,16 @@ static const struct i2c_device_id aic3x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); +static const struct reg_default aic3007_class_d[] = { + /* Class-D speaker driver init; datasheet p. 46 */ + { AIC3X_PAGE_SELECT, 0x0D }, + { 0xD, 0x0D }, + { 0x8, 0x5C }, + { 0x8, 0x5D }, + { 0x8, 0x5C }, + { AIC3X_PAGE_SELECT, 0x00 }, +}; + /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -1463,7 +1444,13 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } - aic3x->control_type = SND_SOC_I2C; + aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap); + if (IS_ERR(aic3x->regmap)) { + ret = PTR_ERR(aic3x->regmap); + return ret; + } + + regcache_cache_only(aic3x->regmap, true); i2c_set_clientdata(i2c, aic3x); if (pdata) { @@ -1533,6 +1520,14 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_gpio; } + if (aic3x->model == AIC3X_MODEL_3007) { + ret = regmap_register_patch(aic3x->regmap, aic3007_class_d, + ARRAY_SIZE(aic3007_class_d)); + if (ret != 0) + dev_err(&i2c->dev, "Failed to init class D: %d\n", + ret); + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); return ret; -- cgit From 19ab2a7a24539d6c80dfe301d2970b075ad3b9ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 11:14:39 +0100 Subject: ASoC: max98088: Set max_register Makes some of the debug functions more useful. Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 31912d59702c..66ceee22fdad 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -552,6 +552,7 @@ static const struct regmap_config max98088_regmap = { .readable_reg = max98088_readable_register, .volatile_reg = max98088_volatile_register, + .max_register = 0xff, .reg_defaults = max98088_reg, .num_reg_defaults = ARRAY_SIZE(max98088_reg), -- cgit From 068416620c0d956b3b382d19dd3000119e280f8c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 18:12:51 +0100 Subject: ASoC: max9850: Convert to direct regmap API usage This prepares for removal of the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 58c38a5b481c..c5dd61785f8d 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -27,18 +28,26 @@ #include "max9850.h" struct max9850_priv { + struct regmap *regmap; unsigned int sysclk; }; /* max9850 register cache */ -static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { - 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +static const struct reg_default max9850_reg[] = { + { 2, 0x0c }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, }; /* these registers are not used at the moment but provided for the sake of * completeness */ -static int max9850_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool max9850_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case MAX9850_STATUSA: @@ -49,6 +58,15 @@ static int max9850_volatile_register(struct snd_soc_codec *codec, } } +static const struct regmap_config max9850_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX9850_DIGITAL_AUDIO, + .volatile_reg = max9850_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + static const unsigned int max9850_tlv[] = { TLV_DB_RANGE_HEAD(4), 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), @@ -225,6 +243,7 @@ static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int max9850_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -234,7 +253,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max9850->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -295,7 +314,7 @@ static int max9850_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -316,10 +335,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .suspend = max9850_suspend, .resume = max9850_resume, .set_bias_level = max9850_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max9850_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max9850_reg, - .volatile_register = max9850_volatile_register, .controls = max9850_controls, .num_controls = ARRAY_SIZE(max9850_controls), @@ -340,6 +355,10 @@ static int max9850_i2c_probe(struct i2c_client *i2c, if (max9850 == NULL) return -ENOMEM; + max9850->regmap = devm_regmap_init_i2c(i2c, &max9850_regmap); + if (IS_ERR(max9850->regmap)) + return PTR_ERR(max9850->regmap); + i2c_set_clientdata(i2c, max9850); ret = snd_soc_register_codec(&i2c->dev, -- cgit From 6f88063c1474b8cdd9254d3934047b6087222145 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:46:05 +0100 Subject: ASoC: cq93vc: Use table based control registration Saves a little code. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index e2c4c0a896e2..e538f4eca980 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -156,10 +156,6 @@ static int cq93vc_probe(struct snd_soc_codec *codec) davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; - /* Set controls */ - snd_soc_add_codec_controls(codec, cq93vc_snd_controls, - ARRAY_SIZE(cq93vc_snd_controls)); - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -180,6 +176,8 @@ static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .probe = cq93vc_probe, .remove = cq93vc_remove, .resume = cq93vc_resume, + .controls = cq93vc_snd_controls, + .num_controls = ARRAY_SIZE(cq93vc_snd_controls), }; static int cq93vc_platform_probe(struct platform_device *pdev) -- cgit From 1201939a6f981cb656872784e39ef443540078cd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:38:16 +0100 Subject: ASoC: cq93vc: Use core I/O functions Support future refactoring by using the core I/O functions rather than calling the driver provided I/O functions directly. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index e538f4eca980..2cbb584b33de 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -64,13 +64,15 @@ static const struct snd_kcontrol_new cq93vc_snd_controls[] = { static int cq93vc_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + u8 reg; if (mute) - cq93vc_write(codec, DAVINCI_VC_REG09, - reg | DAVINCI_VC_REG09_MUTE); + reg = DAVINCI_VC_REG09_MUTE; else - cq93vc_write(codec, DAVINCI_VC_REG09, reg); + reg = 0; + + snd_soc_update_bits(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_MUTE, + reg); return 0; } @@ -97,18 +99,18 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_ON: - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_ON); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_OFF); break; case SND_SOC_BIAS_OFF: /* force all power off */ - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_OFF); break; } -- cgit From d33c33352b3228ca3a422e55981f80fc12dc30f8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:52:41 +0100 Subject: ASoC: cq93vc: Use regmap for I/O Avoid use of the ASoC-specific register I/O functions by converting to use the MMIO regmap provided the core MFD. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 24 +++--------------------- 1 file changed, 3 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 2cbb584b33de..43737a27d79c 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -38,24 +38,6 @@ #include #include -static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct davinci_vc *davinci_vc = codec->control_data; - - return readl(davinci_vc->base + reg); -} - -static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct davinci_vc *davinci_vc = codec->control_data; - - writel(value, davinci_vc->base + reg); - - return 0; -} - static const struct snd_kcontrol_new cq93vc_snd_controls[] = { SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), @@ -156,7 +138,9 @@ static int cq93vc_probe(struct snd_soc_codec *codec) struct davinci_vc *davinci_vc = codec->dev->platform_data; davinci_vc->cq93vc.codec = codec; - codec->control_data = davinci_vc; + codec->control_data = davinci_vc->regmap; + + snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -172,8 +156,6 @@ static int cq93vc_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { - .read = cq93vc_read, - .write = cq93vc_write, .set_bias_level = cq93vc_set_bias_level, .probe = cq93vc_probe, .remove = cq93vc_remove, -- cgit From efeb970ee799b80c984a42d5706081af6047e160 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Sep 2013 23:12:17 -0700 Subject: ASoC: rsnd: remove rsnd_priv_read/write/bset() adg.c only used rsnd_priv_read/write/bset() which is the only user of NULL mod. but, it can be removed. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 10 +++++++--- sound/soc/sh/rcar/rsnd.h | 4 ---- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 2935bbf1811b..9430097979a5 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -21,6 +21,7 @@ struct rsnd_adg { int rate_of_441khz_div_6; int rate_of_48khz_div_6; + u32 ckr; }; #define for_each_rsnd_clk(pos, adg, i) \ @@ -115,6 +116,11 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) found_clock: + /* see rsnd_adg_ssi_clk_init() */ + rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr); + rsnd_mod_write(mod, BRRA, 0x00000002); /* 1/6 */ + rsnd_mod_write(mod, BRRB, 0x00000002); /* 1/6 */ + /* * This "mod" = "ssi" here. * we can get "ssi id" from mod @@ -181,9 +187,7 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) } } - rsnd_priv_bset(priv, SSICKR, 0x00FF0000, ckr); - rsnd_priv_write(priv, BRRA, 0x00000002); /* 1/6 */ - rsnd_priv_write(priv, BRRB, 0x00000002); /* 1/6 */ + adg->ckr = ckr; } int rsnd_adg_probe(struct platform_device *pdev, diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9cc6986a8cfb..3868aaf41cc4 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -78,10 +78,6 @@ struct rsnd_dai_stream; #define rsnd_mod_bset(m, r, s, d) \ rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) -#define rsnd_priv_read(p, r) rsnd_read(p, NULL, RSND_REG_##r) -#define rsnd_priv_write(p, r, d) rsnd_write(p, NULL, RSND_REG_##r, d) -#define rsnd_priv_bset(p, r, s, d) rsnd_bset(p, NULL, RSND_REG_##r, s, d) - u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 data); -- cgit From 55e5b6fd5af04b6d8b0ac6635edf49476ff298ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Sep 2013 23:12:27 -0700 Subject: ASoC: rsnd: use regmap instead of original register mapping method Current Linux kernel is supporting regmap/regmap_field, and, it is good match for Renesas Sound Gen1/Gen2 register mapping. This patch uses regmap instead of original method for register access Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 45 ---------- sound/soc/sh/rcar/gen.c | 224 ++++++++++++++++++++++++++++++----------------- 2 files changed, 143 insertions(+), 126 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a35706028514..fc83f0f2aead 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -106,51 +106,6 @@ (!(priv->info->func) ? -ENODEV : \ priv->info->func(param)) - -/* - * basic function - */ -u32 rsnd_read(struct rsnd_priv *priv, - struct rsnd_mod *mod, enum rsnd_reg reg) -{ - void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); - - BUG_ON(!base); - - return ioread32(base); -} - -void rsnd_write(struct rsnd_priv *priv, - struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data) -{ - void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); - struct device *dev = rsnd_priv_to_dev(priv); - - BUG_ON(!base); - - dev_dbg(dev, "w %p : %08x\n", base, data); - - iowrite32(data, base); -} - -void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 mask, u32 data) -{ - void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); - struct device *dev = rsnd_priv_to_dev(priv); - u32 val; - - BUG_ON(!base); - - val = ioread32(base); - val &= ~mask; - val |= data & mask; - iowrite32(val, base); - - dev_dbg(dev, "s %p : %08x\n", base, val); -} - /* * rsnd_mod functions */ diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 331fc558d796..61212ee97c28 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -24,21 +24,97 @@ struct rsnd_gen_ops { struct rsnd_dai_stream *io); }; -struct rsnd_gen_reg_map { - int index; /* -1 : not supported */ - u32 offset_id; /* offset of ssi0, ssi1, ssi2... */ - u32 offset_adr; /* offset of SSICR, SSISR, ... */ -}; - struct rsnd_gen { void __iomem *base[RSND_BASE_MAX]; - struct rsnd_gen_reg_map reg_map[RSND_REG_MAX]; struct rsnd_gen_ops *ops; + + struct regmap *regmap; + struct regmap_field *regs[RSND_REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) +#define RSND_REG_SET(gen, id, reg_id, offset, _id_offset, _id_size) \ + [id] = { \ + .reg = (unsigned int)gen->base[reg_id] + offset, \ + .lsb = 0, \ + .msb = 31, \ + .id_size = _id_size, \ + .id_offset = _id_offset, \ + } + +/* + * basic function + */ +static int rsnd_regmap_write32(void *context, const void *_data, size_t count) +{ + struct rsnd_priv *priv = context; + struct device *dev = rsnd_priv_to_dev(priv); + u32 *data = (u32 *)_data; + u32 val = data[1]; + void __iomem *reg = (void *)data[0]; + + iowrite32(val, reg); + + dev_dbg(dev, "w %p : %08x\n", reg, val); + + return 0; +} + +static int rsnd_regmap_read32(void *context, + const void *_data, size_t reg_size, + void *_val, size_t val_size) +{ + struct rsnd_priv *priv = context; + struct device *dev = rsnd_priv_to_dev(priv); + u32 *data = (u32 *)_data; + u32 *val = (u32 *)_val; + void __iomem *reg = (void *)data[0]; + + *val = ioread32(reg); + + dev_dbg(dev, "r %p : %08x\n", reg, *val); + + return 0; +} + +static struct regmap_bus rsnd_regmap_bus = { + .write = rsnd_regmap_write32, + .read = rsnd_regmap_read32, + .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, + .val_format_endian_default = REGMAP_ENDIAN_NATIVE, +}; + +u32 rsnd_read(struct rsnd_priv *priv, + struct rsnd_mod *mod, enum rsnd_reg reg) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + u32 val; + + regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + + return val; +} + +void rsnd_write(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); +} + +void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 mask, u32 data) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), + mask, data); +} + /* * Gen2 * will be filled in the future @@ -103,39 +179,64 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv, return ret; } -#define RSND_GEN1_REG_MAP(g, s, i, oi, oa) \ - do { \ - (g)->reg_map[RSND_REG_##i].index = RSND_GEN1_##s; \ - (g)->reg_map[RSND_REG_##i].offset_id = oi; \ - (g)->reg_map[RSND_REG_##i].offset_adr = oa; \ - } while (0) +/* single address mapping */ +#define RSND_GEN1_S_REG(gen, reg, id, offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9) -static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen) +/* multi address mapping */ +#define RSND_GEN1_M_REG(gen, reg, id, offset, _id_offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, _id_offset, 9) + +static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) { - RSND_GEN1_REG_MAP(gen, SRU, SRC_ROUTE_SEL, 0x0, 0x00); - RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL0, 0x0, 0x08); - RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL1, 0x0, 0x0c); - RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL2, 0x0, 0x10); - RSND_GEN1_REG_MAP(gen, SRU, SRC_CTRL, 0x0, 0xc0); - RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE0, 0x0, 0xD0); - RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE1, 0x0, 0xD4); - RSND_GEN1_REG_MAP(gen, SRU, BUSIF_MODE, 0x4, 0x20); - RSND_GEN1_REG_MAP(gen, SRU, BUSIF_ADINR, 0x40, 0x214); - - RSND_GEN1_REG_MAP(gen, ADG, BRRA, 0x0, 0x00); - RSND_GEN1_REG_MAP(gen, ADG, BRRB, 0x0, 0x04); - RSND_GEN1_REG_MAP(gen, ADG, SSICKR, 0x0, 0x08); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL0, 0x0, 0x0c); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL1, 0x0, 0x10); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL3, 0x0, 0x18); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL4, 0x0, 0x1c); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL5, 0x0, 0x20); - - RSND_GEN1_REG_MAP(gen, SSI, SSICR, 0x40, 0x00); - RSND_GEN1_REG_MAP(gen, SSI, SSISR, 0x40, 0x04); - RSND_GEN1_REG_MAP(gen, SSI, SSITDR, 0x40, 0x08); - RSND_GEN1_REG_MAP(gen, SSI, SSIRDR, 0x40, 0x0c); - RSND_GEN1_REG_MAP(gen, SSI, SSIWSR, 0x40, 0x20); + int i; + struct device *dev = rsnd_priv_to_dev(priv); + struct regmap_config regc; + struct reg_field regf[RSND_REG_MAX] = { + RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_SEL, 0x00), + RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), + RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL1, 0x0c), + RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL2, 0x10), + RSND_GEN1_S_REG(gen, SRU, SRC_CTRL, 0xc0), + RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), + RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), + RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4), + RSND_GEN1_M_REG(gen, SRU, BUSIF_ADINR, 0x214, 0x40), + + RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), + RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), + RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), + + RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSITDR, 0x08, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), + }; + + memset(®c, 0, sizeof(regc)); + regc.reg_bits = 32; + regc.val_bits = 32; + + gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); + if (IS_ERR(gen->regmap)) { + dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); + return PTR_ERR(gen->regmap); + } + + for (i = 0; i < RSND_REG_MAX; i++) { + gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); + if (IS_ERR(gen->regs[i])) + return PTR_ERR(gen->regs[i]); + + } + + return 0; } static int rsnd_gen1_probe(struct platform_device *pdev, @@ -147,6 +248,7 @@ static int rsnd_gen1_probe(struct platform_device *pdev, struct resource *sru_res; struct resource *adg_res; struct resource *ssi_res; + int ret; /* * map address @@ -163,7 +265,9 @@ static int rsnd_gen1_probe(struct platform_device *pdev, IS_ERR(gen->base[RSND_GEN1_SSI])) return -ENODEV; - rsnd_gen1_reg_map_init(gen); + ret = rsnd_gen1_regmap_init(priv, gen); + if (ret < 0) + return ret; dev_dbg(dev, "Gen1 device probed\n"); dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start, @@ -210,46 +314,12 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, return gen->ops->path_exit(priv, rdai, io); } -void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, - struct rsnd_mod *mod, - enum rsnd_reg reg) -{ - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - struct device *dev = rsnd_priv_to_dev(priv); - int index; - u32 offset_id, offset_adr; - - if (reg >= RSND_REG_MAX) { - dev_err(dev, "rsnd_reg reg error\n"); - return NULL; - } - - index = gen->reg_map[reg].index; - offset_id = gen->reg_map[reg].offset_id; - offset_adr = gen->reg_map[reg].offset_adr; - - if (index < 0) { - dev_err(dev, "unsupported reg access %d\n", reg); - return NULL; - } - - if (offset_id && mod) - offset_id *= rsnd_mod_id(mod); - - /* - * index/offset were set on gen1/gen2 - */ - - return gen->base[index] + offset_id + offset_adr; -} - int rsnd_gen_probe(struct platform_device *pdev, struct rcar_snd_info *info, struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen; - int i; gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); if (!gen) { @@ -267,14 +337,6 @@ int rsnd_gen_probe(struct platform_device *pdev, priv->gen = gen; - /* - * see - * rsnd_reg_get() - * rsnd_gen_probe() - */ - for (i = 0; i < RSND_REG_MAX; i++) - gen->reg_map[i].index = -1; - return gen->ops->probe(pdev, info, priv); } -- cgit From 4aa11d67b66a84189d25f301e7ef206c4f541692 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 19:26:08 +0100 Subject: ASoC: tlv320aic23: Convert to direct regmap API usage This moves us towards being able to remove the duplicated register I/O code in ASoC. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 54 ++++++++++++++++++++++++------------------ 1 file changed, 31 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 3a6be8c3d557..5d430cc56f51 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -37,11 +38,27 @@ /* * AIC23 register cache */ -static const u16 tlv320aic23_reg[] = { - 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ - 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ - 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +static const struct reg_default tlv320aic23_reg[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x00F9 }, + { 3, 0x00F9 }, + { 4, 0x001A }, + { 5, 0x0004 }, + { 6, 0x0007 }, + { 7, 0x0001 }, + { 8, 0x0020 }, + { 9, 0x0000 }, +}; + +static const struct regmap_config tlv320aic23_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = TLV320AIC23_RESET, + .reg_defaults = tlv320aic23_reg, + .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), + .cache_type = REGCACHE_RBTREE, }; static const char *rec_src_text[] = { "Line", "Mic" }; @@ -171,7 +188,7 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { - enum snd_soc_control_type control_type; + struct regmap *regmap; int mclk; int requested_adc; int requested_dac; @@ -532,7 +549,9 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec) static int tlv320aic23_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); + regcache_mark_dirty(aic23->regmap); + regcache_sync(aic23->regmap); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -540,10 +559,9 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { - struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -552,16 +570,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); - /* Write the register default value to cache for reserved registers, - * so the write to the these registers are suppressed by the cache - * restore code when it skips writes of default registers. - */ - snd_soc_cache_write(codec, 0x0A, 0); - snd_soc_cache_write(codec, 0x0B, 0); - snd_soc_cache_write(codec, 0x0C, 0); - snd_soc_cache_write(codec, 0x0D, 0); - snd_soc_cache_write(codec, 0x0E, 0); - /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -596,9 +604,6 @@ static int tlv320aic23_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { - .reg_cache_size = ARRAY_SIZE(tlv320aic23_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = tlv320aic23_reg, .probe = tlv320aic23_probe, .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, @@ -629,8 +634,11 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, if (aic23 == NULL) return -ENOMEM; + aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap); + if (IS_ERR(aic23->regmap)) + return PTR_ERR(aic23->regmap); + i2c_set_clientdata(i2c, aic23); - aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); -- cgit From 806955dd9cf071ecd99acbaa8c73ae1f34dcf83d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 13:10:33 +0100 Subject: ASoC: tlv320aic26: Convert to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 7b8f3d965f43..32c6b0768e56 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -377,7 +377,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_probe(struct snd_soc_codec *codec) { struct aic26 *aic26 = dev_get_drvdata(codec->dev); - int ret, err, i, reg; + int ret, i, reg; aic26->codec = codec; @@ -403,12 +403,6 @@ static int aic26_probe(struct snd_soc_codec *codec) if (ret) dev_info(codec->dev, "error creating sysfs files\n"); - /* register controls */ - dev_dbg(codec->dev, "Registering controls\n"); - err = snd_soc_add_codec_controls(codec, aic26_snd_controls, - ARRAY_SIZE(aic26_snd_controls)); - WARN_ON(err < 0); - return 0; } @@ -418,6 +412,8 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .write = aic26_reg_write, .reg_cache_size = AIC26_NUM_REGS, .reg_word_size = sizeof(u16), + .controls = aic26_snd_controls, + .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), .dapm_routes = tlv320aic26_dapm_routes, -- cgit From 5b0959d472c215e6d712ac47e64110bd125ddd07 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 13:14:41 +0100 Subject: ASoC: tlv320aic26: Use snd_soc_update_bits() Use snd_soc_update_bits() rather than open coding. Since the register cache is currently only used where update_bits() is used this means the current register cache can be removed entirely. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 51 +++++++++++------------------------------- 1 file changed, 13 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 32c6b0768e56..4d8244750f23 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -74,19 +74,6 @@ static unsigned int aic26_reg_read(struct snd_soc_codec *codec, return value; } -static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return 0; - } - - return cache[reg]; -} - static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -195,19 +182,15 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); - reg &= ~0xf800; if (aic26->master) - reg |= 0x0800; + reg = 0x0800; if (fsref == 48000) - reg |= 0x2000; - snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + reg = 0x2000; + snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL3, 0xf800, reg); /* Audio Control 1 (FSref divisor) */ - reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); - reg &= ~0x0fff; - reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + reg = wlen | aic26->datfm | (divisor << 3) | divisor; + snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL1, 0xfff, reg); return 0; } @@ -219,16 +202,16 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN); + u16 reg; dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n", dai, mute); if (mute) - reg |= 0x8080; + reg = 0x8080; else - reg &= ~0x8080; - snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); + reg = 0; + snd_soc_update_bits(codec, AIC26_REG_DAC_GAIN, 0x8000, reg); return 0; } @@ -346,7 +329,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = snd_soc_read(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -360,11 +343,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, const char *buf, size_t count) { struct aic26 *aic26 = dev_get_drvdata(dev); - int val; - val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); - val |= 0x8000; - snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_update_bits(aic26->codec, AIC26_REG_AUDIO_CTRL2, + 0x8000, 0x800); return count; } @@ -377,7 +358,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_probe(struct snd_soc_codec *codec) { struct aic26 *aic26 = dev_get_drvdata(codec->dev); - int ret, i, reg; + int ret, reg; aic26->codec = codec; @@ -393,10 +374,6 @@ static int aic26_probe(struct snd_soc_codec *codec) reg |= 0x0800; /* set master mode */ snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); - /* Fill register cache */ - for (i = 0; i < codec->driver->reg_cache_size; i++) - snd_soc_read(codec, i); - /* Register the sysfs files for debugging */ /* Create SysFS files */ ret = device_create_file(codec->dev, &dev_attr_keyclick); @@ -410,8 +387,6 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .probe = aic26_probe, .read = aic26_reg_read, .write = aic26_reg_write, - .reg_cache_size = AIC26_NUM_REGS, - .reg_word_size = sizeof(u16), .controls = aic26_snd_controls, .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, -- cgit From 7fbdeb809050cb958f3baa83dcc643f9a2f287f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 13:29:44 +0100 Subject: ASoC: tlv320aic26: Convert to direct regmap API usage This moves us towards being able to remove the duplicated register I/O code in ASoC. The datasheet and the driver document the device as having a register map divided into pages but since the paging is actually done by sending the page address and the register address with each transaction this is no different to having a simple register address. The datasheet does also document the low five bits of the 16 bit "command" as unused which we could represent as padding but it seems simpler and less confusing to things that use block transfers or autoincrement to represent these as part of the register address. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 80 +++++++----------------------------------- sound/soc/codecs/tlv320aic26.h | 5 +-- 2 files changed, 13 insertions(+), 72 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 4d8244750f23..94a658fa6d97 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,6 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; + struct regmap *regmap; struct snd_soc_codec *codec; int master; int datfm; @@ -40,72 +41,6 @@ struct aic26 { int keyclick_len; }; -/* --------------------------------------------------------------------- - * Register access routines - */ -static unsigned int aic26_reg_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - u16 cmd, value; - u8 buffer[2]; - int rc; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return 0; - } - - /* Do SPI transfer; first 16bits are command; remaining is - * register contents */ - cmd = AIC26_READ_COMMAND_WORD(reg); - buffer[0] = (cmd >> 8) & 0xff; - buffer[1] = cmd & 0xff; - rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2); - if (rc) { - dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); - return -EIO; - } - value = (buffer[0] << 8) | buffer[1]; - - /* Update the cache before returning with the value */ - cache[reg] = value; - return value; -} - -static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - u16 cmd; - u8 buffer[4]; - int rc; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return -EINVAL; - } - - /* Do SPI transfer; first 16bits are command; remaining is data - * to write into register */ - cmd = AIC26_WRITE_COMMAND_WORD(reg); - buffer[0] = (cmd >> 8) & 0xff; - buffer[1] = cmd & 0xff; - buffer[2] = value >> 8; - buffer[3] = value; - rc = spi_write(aic26->spi, buffer, 4); - if (rc) { - dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); - return -EIO; - } - - /* update cache before returning */ - cache[reg] = value; - return 0; -} - static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICIN"), SND_SOC_DAPM_INPUT("AUX"), @@ -360,6 +295,8 @@ static int aic26_probe(struct snd_soc_codec *codec) struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, reg; + snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + aic26->codec = codec; /* Reset the codec to power on defaults */ @@ -385,8 +322,6 @@ static int aic26_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver aic26_soc_codec_dev = { .probe = aic26_probe, - .read = aic26_reg_read, - .write = aic26_reg_write, .controls = aic26_snd_controls, .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, @@ -395,6 +330,11 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), }; +static const struct regmap_config aic26_regmap = { + .reg_bits = 16, + .val_bits = 16, +}; + /* --------------------------------------------------------------------- * SPI device portion of driver: probe and release routines and SPI * driver registration. @@ -411,6 +351,10 @@ static int aic26_spi_probe(struct spi_device *spi) if (!aic26) return -ENOMEM; + aic26->regmap = devm_regmap_init_spi(spi, &aic26_regmap); + if (IS_ERR(aic26->regmap)) + return PTR_ERR(aic26->regmap); + /* Initialize the driver data */ aic26->spi = spi; dev_set_drvdata(&spi->dev, aic26); diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 67f19c3bebe6..629b85e75409 100644 --- a/sound/soc/codecs/tlv320aic26.h +++ b/sound/soc/codecs/tlv320aic26.h @@ -9,10 +9,7 @@ #define _TLV320AIC16_H_ /* AIC26 Registers */ -#define AIC26_READ_COMMAND_WORD(addr) ((1 << 15) | (addr << 5)) -#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5)) -#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) -#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) +#define AIC26_PAGE_ADDR(page, offset) ((page << 11) | offset << 5) /* Page 0: Auxiliary data registers */ #define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) -- cgit From c29c2d4e6ba4f8f5868a4c6dd75a4c23e1993721 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:14:31 +0100 Subject: mfd: mc13xxx: Don't require lock for simple register I/O Since the conversion to regmap there has been no need for device level locking for I/O as regmap provides locking so remove the locks. Signed-off-by: Mark Brown Signed-off-by: Lee Jones --- drivers/mfd/mc13xxx-core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 2a9b100c4825..dbbf8ee3f592 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -158,8 +158,6 @@ int mc13xxx_reg_read(struct mc13xxx *mc13xxx, unsigned int offset, u32 *val) { int ret; - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - if (offset > MC13XXX_NUMREGS) return -EINVAL; @@ -172,8 +170,6 @@ EXPORT_SYMBOL(mc13xxx_reg_read); int mc13xxx_reg_write(struct mc13xxx *mc13xxx, unsigned int offset, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x\n", offset, val); if (offset > MC13XXX_NUMREGS || val > 0xffffff) @@ -186,7 +182,6 @@ EXPORT_SYMBOL(mc13xxx_reg_write); int mc13xxx_reg_rmw(struct mc13xxx *mc13xxx, unsigned int offset, u32 mask, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); BUG_ON(val & ~mask); dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x (mask: 0x%06x)\n", offset, val, mask); -- cgit From fd792f8fbcfa95674b6c417429f576ad1d808086 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:14:32 +0100 Subject: mfd: mc13xxx: Move SPI erratum workaround into SPI I/O function Move the workaround for double sending AUDIO_CODEC and AUDIO_DAC writes into the SPI core, aiding refactoring to eliminate the ASoC custom I/O functions and avoiding the extra writes for I2C. Signed-off-by: Mark Brown Signed-off-by: Lee Jones --- drivers/mfd/mc13xxx-spi.c | 5 +++++ include/linux/mfd/mc13xxx.h | 7 +++++++ sound/soc/codecs/mc13783.c | 4 ---- 3 files changed, 12 insertions(+), 4 deletions(-) diff --git a/drivers/mfd/mc13xxx-spi.c b/drivers/mfd/mc13xxx-spi.c index 77189daadf1e..5f14ef6693c2 100644 --- a/drivers/mfd/mc13xxx-spi.c +++ b/drivers/mfd/mc13xxx-spi.c @@ -94,10 +94,15 @@ static int mc13xxx_spi_write(void *context, const void *data, size_t count) { struct device *dev = context; struct spi_device *spi = to_spi_device(dev); + const char *reg = data; if (count != 4) return -ENOTSUPP; + /* include errata fix for spi audio problems */ + if (*reg == MC13783_AUDIO_CODEC || *reg == MC13783_AUDIO_DAC) + spi_write(spi, data, count); + return spi_write(spi, data, count); } diff --git a/include/linux/mfd/mc13xxx.h b/include/linux/mfd/mc13xxx.h index 41ed59276c00..67c17b5a6f44 100644 --- a/include/linux/mfd/mc13xxx.h +++ b/include/linux/mfd/mc13xxx.h @@ -41,6 +41,13 @@ int mc13xxx_adc_do_conversion(struct mc13xxx *mc13xxx, unsigned int mode, unsigned int channel, u8 ato, bool atox, unsigned int *sample); +#define MC13783_AUDIO_RX0 36 +#define MC13783_AUDIO_RX1 37 +#define MC13783_AUDIO_TX 38 +#define MC13783_SSI_NETWORK 39 +#define MC13783_AUDIO_CODEC 40 +#define MC13783_AUDIO_DAC 41 + #define MC13XXX_IRQ_ADCDONE 0 #define MC13XXX_IRQ_ADCBISDONE 1 #define MC13XXX_IRQ_TS 2 diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index ea141e1d6f28..4d3c8fd8c5db 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -125,10 +125,6 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - /* include errata fix for spi audio problems */ - if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) - ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - mc13xxx_unlock(priv->mc13xxx); return ret; -- cgit From 2d9215c1ecd6f133952bc081a288dbb180816290 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Sep 2013 19:04:17 +0100 Subject: ASoC: mc13783: Use regmap directly from ASoC As part of a push to remove the register I/O functionality from ASoC (since it is now duplicated in the regmap API) convert the mc13783 driver to use regmap directly. Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 55 ++++++++-------------------------------------- 1 file changed, 9 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 4d3c8fd8c5db..eedbf05b8e96 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -30,16 +30,10 @@ #include #include #include +#include #include "mc13783.h" -#define MC13783_AUDIO_RX0 36 -#define MC13783_AUDIO_RX1 37 -#define MC13783_AUDIO_TX 38 -#define MC13783_SSI_NETWORK 39 -#define MC13783_AUDIO_CODEC 40 -#define MC13783_AUDIO_DAC 41 - #define AUDIO_RX0_ALSPEN (1 << 5) #define AUDIO_RX0_ALSPSEL (1 << 7) #define AUDIO_RX0_ADDCDC (1 << 21) @@ -95,41 +89,12 @@ struct mc13783_priv { struct mc13xxx *mc13xxx; + struct regmap *regmap; enum mc13783_ssi_port adc_ssi_port; enum mc13783_ssi_port dac_ssi_port; }; -static unsigned int mc13783_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - unsigned int value = 0; - - mc13xxx_lock(priv->mc13xxx); - - mc13xxx_reg_read(priv->mc13xxx, reg, &value); - - mc13xxx_unlock(priv->mc13xxx); - - return value; -} - -static int mc13783_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; - - mc13xxx_lock(priv->mc13xxx); - - ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - - mc13xxx_unlock(priv->mc13xxx); - - return ret; -} - /* Mapping between sample rates and register value */ static unsigned int mc13783_rates[] = { 8000, 11025, 12000, 16000, @@ -583,8 +548,14 @@ static struct snd_kcontrol_new mc13783_control_list[] = { static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; - mc13xxx_lock(priv->mc13xxx); + codec->control_data = dev_get_regmap(codec->dev->parent, NULL); + ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* these are the reset values */ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893); @@ -608,8 +579,6 @@ static int mc13783_probe(struct snd_soc_codec *codec) mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, 0, AUDIO_SSI_SEL); - mc13xxx_unlock(priv->mc13xxx); - return 0; } @@ -617,13 +586,9 @@ static int mc13783_remove(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - mc13xxx_lock(priv->mc13xxx); - /* Make sure VAUDIOON is off */ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0); - mc13xxx_unlock(priv->mc13xxx); - return 0; } @@ -713,8 +678,6 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { .probe = mc13783_probe, .remove = mc13783_remove, - .read = mc13783_read, - .write = mc13783_write, .controls = mc13783_control_list, .num_controls = ARRAY_SIZE(mc13783_control_list), .dapm_widgets = mc13783_dapm_widgets, -- cgit From e0f17c75d9352c38926da1b4d8dbefc2d9942006 Mon Sep 17 00:00:00 2001 From: Peter Senna Tschudin Date: Sun, 22 Sep 2013 20:44:12 +0200 Subject: ALSA: Fix assignment of 0/1 to bool variables Convert 0 to false and 1 to true when assigning values to bool variables. Inspired by commit 3db1cd5c05f35fb43eb134df6f321de4e63141f2. The simplified semantic patch that find this problem is as follows (http://coccinelle.lip6.fr/): @@ bool b; @@ ( -b = 0 +b = false | -b = 1 +b = true ) Signed-off-by: Peter Senna Tschudin Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 14 +++++++------- sound/usb/mixer.c | 4 ++-- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index c8e121611593..1aef7128f7ca 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -715,14 +715,14 @@ snd_azf3328_mixer_ac97_read(struct snd_ac97 *ac97, unsigned short reg_ac97) const struct snd_azf3328 *chip = ac97->private_data; unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); unsigned short reg_val = 0; - bool unsupported = 0; + bool unsupported = false; snd_azf3328_dbgmixer( "snd_azf3328_mixer_ac97_read reg_ac97 %u\n", reg_ac97 ); if (reg_azf & AZF_AC97_REG_UNSUPPORTED) - unsupported = 1; + unsupported = true; else { if (reg_azf & AZF_AC97_REG_REAL_IO_READ) reg_val = snd_azf3328_mixer_inw(chip, @@ -759,7 +759,7 @@ snd_azf3328_mixer_ac97_read(struct snd_ac97 *ac97, unsigned short reg_ac97) reg_val = azf_emulated_ac97_vendor_id & 0xffff; break; default: - unsupported = 1; + unsupported = true; break; } } @@ -776,14 +776,14 @@ snd_azf3328_mixer_ac97_write(struct snd_ac97 *ac97, { const struct snd_azf3328 *chip = ac97->private_data; unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); - bool unsupported = 0; + bool unsupported = false; snd_azf3328_dbgmixer( "snd_azf3328_mixer_ac97_write reg_ac97 %u val %u\n", reg_ac97, val ); if (reg_azf & AZF_AC97_REG_UNSUPPORTED) - unsupported = 1; + unsupported = true; else { if (reg_azf & AZF_AC97_REG_REAL_IO_WRITE) snd_azf3328_mixer_outw( @@ -808,7 +808,7 @@ snd_azf3328_mixer_ac97_write(struct snd_ac97 *ac97, */ break; default: - unsupported = 1; + unsupported = true; break; } } @@ -1559,7 +1559,7 @@ snd_azf3328_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_azf3328_codec_data *codec = runtime->private_data; int result = 0; u16 flags1; - bool previously_muted = 0; + bool previously_muted = false; bool is_main_mixer_playback_codec = (AZF_CODEC_PLAYBACK == codec->type); snd_azf3328_dbgcalls("snd_azf3328_pcm_trigger cmd %d\n", cmd); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 95558ef4a7a0..44b0ba4feab3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1151,14 +1151,14 @@ static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, const char *names_to_check[] = { "Headset", "headset", "Headphone", "headphone", NULL}; const char **s; - bool found = 0; + bool found = false; if (strcmp("Speaker", kctl->id.name)) return; for (s = names_to_check; *s; s++) if (strstr(card->shortname, *s)) { - found = 1; + found = true; break; } -- cgit From 976b6c064a957445eb0573b270f2d0282630e9b9 Mon Sep 17 00:00:00 2001 From: Alan Stern Date: Tue, 24 Sep 2013 15:51:58 -0400 Subject: ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern CC: Clemens Ladisch Tested-by: Daniel Mack Tested-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/card.c | 8 ---- sound/usb/card.h | 10 +++-- sound/usb/endpoint.c | 106 +++++++++++++++++++++++++++++---------------------- sound/usb/endpoint.h | 2 + sound/usb/pcm.c | 14 ++++++- sound/usb/usbaudio.h | 1 - 6 files changed, 82 insertions(+), 59 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 64952e2d3ed1..d1f54dfe41d5 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -79,7 +79,6 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card * /* Vendor/product IDs for this card */ static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; -static int nrpacks = 8; /* max. number of packets per urb */ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; @@ -94,8 +93,6 @@ module_param_array(vid, int, NULL, 0444); MODULE_PARM_DESC(vid, "Vendor ID for the USB audio device."); module_param_array(pid, int, NULL, 0444); MODULE_PARM_DESC(pid, "Product ID for the USB audio device."); -module_param(nrpacks, int, 0644); -MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB."); module_param_array(device_setup, int, NULL, 0444); MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); module_param(ignore_ctl_error, bool, 0444); @@ -374,7 +371,6 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, chip->dev = dev; chip->card = card; chip->setup = device_setup[idx]; - chip->nrpacks = nrpacks; chip->autoclock = autoclock; chip->probing = 1; @@ -756,10 +752,6 @@ static struct usb_driver usb_audio_driver = { static int __init snd_usb_audio_init(void) { - if (nrpacks < 1 || nrpacks > MAX_PACKS) { - printk(KERN_WARNING "invalid nrpacks value.\n"); - return -EINVAL; - } return usb_register(&usb_audio_driver); } diff --git a/sound/usb/card.h b/sound/usb/card.h index 5ecacaa90b53..ca98a9b915c9 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -2,11 +2,11 @@ #define __USBAUDIO_CARD_H #define MAX_NR_RATES 1024 -#define MAX_PACKS 20 +#define MAX_PACKS 6 /* per URB */ #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ -#define MAX_URBS 8 +#define MAX_URBS 12 #define SYNC_URBS 4 /* always four urbs for sync */ -#define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ +#define MAX_QUEUE 18 /* try not to exceed this queue length, in ms */ struct audioformat { struct list_head list; @@ -87,6 +87,7 @@ struct snd_usb_endpoint { unsigned int phase; /* phase accumulator */ unsigned int maxpacksize; /* max packet size in bytes */ unsigned int maxframesize; /* max packet size in frames */ + unsigned int max_urb_frames; /* max URB size in frames */ unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int syncmaxsize; /* sync endpoint packet size */ @@ -116,6 +117,8 @@ struct snd_usb_substream { unsigned int channels_max; /* max channels in the all audiofmts */ unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ + unsigned int period_frames; /* current frames per period */ + unsigned int buffer_periods; /* current periods per buffer */ unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ @@ -125,6 +128,7 @@ struct snd_usb_substream { unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ + unsigned int frame_limit; /* limits number of packets in URB */ /* data and sync endpoints for this stream */ unsigned int ep_num; /* the endpoint number */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 93e970f2b3c0..21dc6422d747 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -574,11 +574,14 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) { - unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms; - int is_playback = usb_pipeout(ep->pipe); + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs, i; int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { @@ -611,58 +614,67 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, else ep->curpacksize = maxsize; - if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { packs_per_ms = 8 >> ep->datainterval; - else - packs_per_ms = 1; - - if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { - urb_packs = max(ep->chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int) MAX_PACKS); + max_packs_per_urb = MAX_PACKS_HS; } else { - urb_packs = 1; + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; } + if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); - urb_packs *= packs_per_ms; + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { - if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) - urb_packs = min(urb_packs, 1U << sync_ep->syncinterval); + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, packs_per_ms); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ep->nurbs = MAX_URBS; - /* decide how many packets to be used */ - if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { - unsigned int minsize, maxpacks; + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { /* determine how small a packet can be */ - minsize = (ep->freqn >> (16 - ep->datainterval)) - * (frame_bits >> 3); + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); /* with sync from device, assume it can be 12% lower */ if (sync_ep) minsize -= minsize >> 3; minsize = max(minsize, 1u); - total_packs = (period_bytes + minsize - 1) / minsize; - /* we need at least two URBs for queueing */ - if (total_packs < 2) { - total_packs = 2; - } else { - /* and we don't want too long a queue either */ - maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); - total_packs = min(total_packs, maxpacks); - } - } else { - while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) - urb_packs >>= 1; - total_packs = MAX_URBS * urb_packs; - } - ep->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (ep->nurbs > MAX_URBS) { - /* too much... */ - ep->nurbs = MAX_URBS; - total_packs = MAX_URBS * urb_packs; - } else if (ep->nurbs < 2) { - /* too little - we need at least two packets - * to ensure contiguous playback/capture - */ - ep->nurbs = 2; + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ep->max_urb_frames = DIV_ROUND_UP(frames_per_period, + urbs_per_period); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ep->nurbs = min(max_urbs, urbs_per_period * periods_per_buffer); } /* allocate and initialize data urbs */ @@ -670,8 +682,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, struct snd_urb_ctx *u = &ep->urb[i]; u->index = i; u->ep = ep; - u->packets = (i + 1) * total_packs / ep->nurbs - - i * total_packs / ep->nurbs; + u->packets = urb_packs; u->buffer_size = maxsize * u->packets; if (fmt->fmt_type == UAC_FORMAT_TYPE_II) @@ -748,6 +759,8 @@ out_of_memory: * @pcm_format: the audio fomat. * @channels: the number of audio channels. * @period_bytes: the number of bytes in one alsa period. + * @period_frames: the number of frames in one alsa period. + * @buffer_periods: the number of periods in one alsa buffer. * @rate: the frame rate. * @fmt: the USB audio format information * @sync_ep: the sync endpoint to use, if any @@ -760,6 +773,8 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int period_frames, + unsigned int buffer_periods, unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) @@ -793,7 +808,8 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, switch (ep->type) { case SND_USB_ENDPOINT_TYPE_DATA: err = data_ep_set_params(ep, pcm_format, channels, - period_bytes, fmt, sync_ep); + period_bytes, period_frames, + buffer_periods, fmt, sync_ep); break; case SND_USB_ENDPOINT_TYPE_SYNC: err = sync_ep_set_params(ep, fmt); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 2287adf5ca59..3bd02f0d226c 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -12,6 +12,8 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int period_frames, + unsigned int buffer_periods, unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b375d58871e7..19e79953f2e3 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -595,6 +595,7 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) subs->pcm_format, subs->channels, subs->period_bytes, + 0, 0, subs->cur_rate, subs->cur_audiofmt, NULL); @@ -631,6 +632,7 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) subs->pcm_format, sync_fp->channels, sync_period_bytes, + 0, 0, subs->cur_rate, sync_fp, NULL); @@ -653,6 +655,8 @@ static int configure_endpoint(struct snd_usb_substream *subs) subs->pcm_format, subs->channels, subs->period_bytes, + subs->period_frames, + subs->buffer_periods, subs->cur_rate, subs->cur_audiofmt, subs->sync_endpoint); @@ -689,6 +693,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); + subs->period_frames = params_period_size(hw_params); + subs->buffer_periods = params_periods(hw_params); subs->channels = params_channels(hw_params); subs->cur_rate = params_rate(hw_params); @@ -1363,6 +1369,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, frames = 0; urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); + subs->frame_limit += ep->max_urb_frames; for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; @@ -1377,6 +1384,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->transfer_done += counts; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; + subs->frame_limit = 0; period_elapsed = 1; if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { @@ -1399,8 +1407,10 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, break; } } - if (period_elapsed && - !snd_usb_endpoint_implicit_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ + /* finish at the period boundary or after enough frames */ + if ((period_elapsed || + subs->transfer_done >= subs->frame_limit) && + !snd_usb_endpoint_implicit_feedback_sink(ep)) break; } bytes = frames * ep->stride; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index caabe9b3af49..5d2fe0530745 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -55,7 +55,6 @@ struct snd_usb_audio { struct list_head mixer_list; /* list of mixer interfaces */ int setup; /* from the 'device_setup' module param */ - int nrpacks; /* from the 'nrpacks' module param */ bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ -- cgit From e84841f9ba134d3aa4cad5c16d05712672583c76 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 14 Sep 2013 00:35:47 +0900 Subject: ALSA: firewire-lib: use inlune function to calculate frame bytes Calculating frame bytes can be replaced with inline function in include/sound/pcm.h. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index ea995af6d049..4b08b25a4db8 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -268,7 +268,7 @@ static void amdtp_write_s32(struct amdtp_out_stream *s, channels = s->pcm_channels; src = (void *)runtime->dma_area + - s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frames_to_bytes(runtime, s->pcm_buffer_pointer); remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; frame_step = s->data_block_quadlets - channels; @@ -294,7 +294,7 @@ static void amdtp_write_s16(struct amdtp_out_stream *s, channels = s->pcm_channels; src = (void *)runtime->dma_area + - s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frames_to_bytes(runtime, s->pcm_buffer_pointer); remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; frame_step = s->data_block_quadlets - channels; -- cgit From 83cbe35b874621a23ca468621c0d833b76a1b8de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 19:22:54 +0100 Subject: ASoC: sn95031: Convert to regmap This moves us towards being able to remove the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index dba26e63844e..13045f2af4d3 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -164,30 +164,28 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) } /*end - adc helper functions */ -static inline unsigned int sn95031_read(struct snd_soc_codec *codec, - unsigned int reg) +static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val) { u8 value = 0; int ret; ret = intel_scu_ipc_ioread8(reg, &value); - if (ret) - pr_err("read of %x failed, err %d\n", reg, ret); - return value; + if (ret == 0) + *val = value; + return ret; } -static inline int sn95031_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int sn95031_write(void *ctx, unsigned int reg, unsigned int value) { - int ret; - - ret = intel_scu_ipc_iowrite8(reg, value); - if (ret) - pr_err("write of %x failed, err %d\n", reg, ret); - return ret; + return intel_scu_ipc_iowrite8(reg, value); } +static const struct regmap_config sn95031_regmap = { + .reg_read = sn95031_read, + .reg_write = sn95031_write, +}; + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -827,6 +825,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -886,8 +886,6 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, - .read = sn95031_read, - .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, @@ -898,7 +896,14 @@ static struct snd_soc_codec_driver sn95031_codec = { static int sn95031_device_probe(struct platform_device *pdev) { + struct regmap *regmap; + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + + regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + return snd_soc_register_codec(&pdev->dev, &sn95031_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } -- cgit From 752b776435cb35da27a0bbec8deecc33b3461288 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 11:36:26 +0100 Subject: ASoC: tlv320aic32x4: Move GPIO acquisition to I2C probe This is more idiomatic and interacts better with deferred probe. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 2ed57d4aa445..cf70bf86c344 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -617,16 +617,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - int ret; codec->hw_write = (hw_write_t) i2c_master_send; codec->control_data = aic32x4->control_data; if (aic32x4->rstn_gpio >= 0) { - ret = devm_gpio_request_one(codec->dev, aic32x4->rstn_gpio, - GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); - if (ret != 0) - return ret; ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); } @@ -735,6 +730,13 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } + if (aic32x4->rstn_gpio >= 0) { + ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); + if (ret != 0) + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); return ret; -- cgit From 4d208ca429ad424595fd08c0cca323605ebfc38b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 11:37:53 +0100 Subject: ASoC: tlv320aic32x4: Convert to direct regmap API usage This moves us towards being able to remove the duplicate register I/O functionality in ASoC and saves some code. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 89 +++++++++++----------------------------- 1 file changed, 23 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index cf70bf86c344..18cdcca9014c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -60,9 +60,8 @@ struct aic32x4_rate_divs { }; struct aic32x4_priv { + struct regmap *regmap; u32 sysclk; - u8 page_no; - void *control_data; u32 power_cfg; u32 micpga_routing; bool swapdacs; @@ -262,67 +261,25 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"Right ADC", NULL, "Right Input Mixer"}, }; -static inline int aic32x4_change_page(struct snd_soc_codec *codec, - unsigned int new_page) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - u8 data[2]; - int ret; - - data[0] = 0x00; - data[1] = new_page & 0xff; - - ret = codec->hw_write(codec->control_data, data, 2); - if (ret == 2) { - aic32x4->page_no = new_page; - return 0; - } else { - return ret; - } -} - -static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - unsigned int page = reg / 128; - unsigned int fixed_reg = reg % 128; - u8 data[2]; - int ret; - - /* A write to AIC32X4_PSEL is really a non-explicit page change */ - if (reg == AIC32X4_PSEL) - return aic32x4_change_page(codec, val); - - if (aic32x4->page_no != page) { - ret = aic32x4_change_page(codec, page); - if (ret != 0) - return ret; - } +static const struct regmap_range_cfg aic32x4_regmap_pages[] = { + { + .selector_reg = 0, + .selector_mask = 0xff, + .window_start = 0, + .window_len = 128, + .range_min = AIC32X4_PAGE1, + .range_max = AIC32X4_PAGE1 + 127, + }, +}; - data[0] = fixed_reg & 0xff; - data[1] = val & 0xff; +static const struct regmap_config aic32x4_regmap = { + .reg_bits = 8, + .val_bits = 8, - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - unsigned int page = reg / 128; - unsigned int fixed_reg = reg % 128; - int ret; - - if (aic32x4->page_no != page) { - ret = aic32x4_change_page(codec, page); - if (ret != 0) - return ret; - } - return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); -} + .max_register = AIC32X4_RMICPGAVOL, + .ranges = aic32x4_regmap_pages, + .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), +}; static inline int aic32x4_get_divs(int mclk, int rate) { @@ -618,8 +575,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - codec->hw_write = (hw_write_t) i2c_master_send; - codec->control_data = aic32x4->control_data; + snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (aic32x4->rstn_gpio >= 0) { ndelay(10); @@ -687,8 +643,6 @@ static int aic32x4_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { - .read = aic32x4_read, - .write = aic32x4_write, .probe = aic32x4_probe, .remove = aic32x4_remove, .suspend = aic32x4_suspend, @@ -715,7 +669,10 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, if (aic32x4 == NULL) return -ENOMEM; - aic32x4->control_data = i2c; + aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); + if (IS_ERR(aic32x4->regmap)) + return PTR_ERR(aic32x4->regmap); + i2c_set_clientdata(i2c, aic32x4); if (pdata) { -- cgit From ce3d060990cd799cef4eeffc290090fb5da15e94 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Thu, 19 Sep 2013 11:20:43 +0200 Subject: ASoc: kirkwood: Extend the min and max number of bytes per period This patch extends the min and max number of bytes per period. It mainly permits to reduce the sound delay in MIDI real-time playing. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index f8e1ccc1c58c..bf23afbba1d7 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -123,8 +123,8 @@ /* need to find where they come from */ #define KIRKWOOD_SND_MIN_PERIODS 8 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000 -#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) -- cgit From 3a429eea10ded31d2ff088432d02072165a099f1 Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Fri, 20 Sep 2013 15:54:54 -0700 Subject: ASoC: atmel-pcm: fix warning i386 allmodconfig: sound/soc/atmel/atmel-pcm.c: In function 'atmel_pcm_preallocate_dma_buffer': sound/soc/atmel/atmel-pcm.c:52: warning: cast to pointer from integer of different size Signed-off-by: Andrew Morton Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 3109db7b9017..612e5801003f 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -50,7 +50,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", - (void *)buf->area, (void *)buf->addr, size); + (void *)buf->area, (void *)(long)buf->addr, size); if (!buf->area) return -ENOMEM; -- cgit From d60336e2f136287de821901d4a1b56179a0f7b69 Mon Sep 17 00:00:00 2001 From: Chen Gang Date: Mon, 23 Sep 2013 11:36:21 +0800 Subject: ASoC: fsl_ssi: let check zero instead of check NO_IRQ NO_IRQ may be defined as '(unsigned int) -1' in some architectures (arm, sh ...), and either may not be defined in some architectures which can enable SND_SOC_FSL_SSI (e.g. allmodconfig for arc). When irq_of_parse_and_map() fails, it will always return 0, so need check zero instead of NO_IRQ, or will cause compiling issue or run time bug in some architectures. Signed-off-by: Chen Gang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index cdbb641ef518..35e277379b86 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -929,7 +929,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); - if (ssi_private->irq == NO_IRQ) { + if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; } -- cgit From d6173df35f2dbd0e11f2361fc979ebf2e53cb6cc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Sep 2013 19:36:11 +0100 Subject: ASoC: si476x: Remove custom register I/O implementation The current si476x I/O implementation wraps the regmap for the core with functions that make the register map cache only when the device is powered down. This implementation appears to be incomplete since there is no code to synchronise the cache so writes done while the core is powered down will be ignored, the device will only be configured if it is powered. A better and more idiomatic approach would be to have the MFD manage the cache, making the device cache only when it powers things down. This also allows ASoC to use the standard regmap helpers for the device which helps remove the ASoC custom ones so do convert to do that. Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 46 +--------------------------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38f3b105c17d..03645ce42063 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -60,48 +60,6 @@ enum si476x_pcm_format { SI476X_PCM_FORMAT_S24_LE = 6, }; -static unsigned int si476x_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - int err; - unsigned int val; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_read(core->regmap, reg, &val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - if (err < 0) - return err; - - return val; -} - -static int si476x_codec_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val) -{ - int err; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_write(core->regmap, reg, val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - return err; -} - static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), @@ -239,7 +197,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = i2c_mfd_cell_to_core(codec->dev); + codec->control_data = dev_get_regmap(codec->dev->parent, NULL); return 0; } @@ -268,8 +226,6 @@ static struct snd_soc_dai_driver si476x_dai = { static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, - .read = si476x_codec_read, - .write = si476x_codec_write, .dapm_widgets = si476x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), .dapm_routes = si476x_dapm_routes, -- cgit From c3df37c9380d70f19a9cb2de4c7d58d7822a4b35 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 13:47:08 +0200 Subject: ASoC: adau1373: Convert to direct regmap usage Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 295 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 235 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1aa10ddf3a61..c57c1f81a611 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -32,6 +32,7 @@ struct adau1373_dai { }; struct adau1373 { + struct regmap *regmap; struct adau1373_dai dais[3]; }; @@ -152,37 +153,172 @@ struct adau1373 { #define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 #define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 -static const uint8_t adau1373_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ - 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ - 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ - 0x00, 0x1f, 0x0f, 0x00, 0x00, +static const struct reg_default adau1373_reg_defaults[] = { + { ADAU1373_INPUT_MODE, 0x00 }, + { ADAU1373_AINL_CTRL(0), 0x00 }, + { ADAU1373_AINR_CTRL(0), 0x00 }, + { ADAU1373_AINL_CTRL(1), 0x00 }, + { ADAU1373_AINR_CTRL(1), 0x00 }, + { ADAU1373_AINL_CTRL(2), 0x00 }, + { ADAU1373_AINR_CTRL(2), 0x00 }, + { ADAU1373_AINL_CTRL(3), 0x00 }, + { ADAU1373_AINR_CTRL(3), 0x00 }, + { ADAU1373_LLINE_OUT(0), 0x00 }, + { ADAU1373_RLINE_OUT(0), 0x00 }, + { ADAU1373_LLINE_OUT(1), 0x00 }, + { ADAU1373_RLINE_OUT(1), 0x00 }, + { ADAU1373_LSPK_OUT, 0x00 }, + { ADAU1373_RSPK_OUT, 0x00 }, + { ADAU1373_LHP_OUT, 0x00 }, + { ADAU1373_RHP_OUT, 0x00 }, + { ADAU1373_ADC_GAIN, 0x00 }, + { ADAU1373_LADC_MIXER, 0x00 }, + { ADAU1373_RADC_MIXER, 0x00 }, + { ADAU1373_LLINE1_MIX, 0x00 }, + { ADAU1373_RLINE1_MIX, 0x00 }, + { ADAU1373_LLINE2_MIX, 0x00 }, + { ADAU1373_RLINE2_MIX, 0x00 }, + { ADAU1373_LSPK_MIX, 0x00 }, + { ADAU1373_RSPK_MIX, 0x00 }, + { ADAU1373_LHP_MIX, 0x00 }, + { ADAU1373_RHP_MIX, 0x00 }, + { ADAU1373_EP_MIX, 0x00 }, + { ADAU1373_HP_CTRL, 0x00 }, + { ADAU1373_HP_CTRL2, 0x00 }, + { ADAU1373_LS_CTRL, 0x00 }, + { ADAU1373_EP_CTRL, 0x00 }, + { ADAU1373_MICBIAS_CTRL1, 0x00 }, + { ADAU1373_MICBIAS_CTRL2, 0x00 }, + { ADAU1373_OUTPUT_CTRL, 0x00 }, + { ADAU1373_PWDN_CTRL1, 0x00 }, + { ADAU1373_PWDN_CTRL2, 0x00 }, + { ADAU1373_PWDN_CTRL3, 0x00 }, + { ADAU1373_DPLL_CTRL(0), 0x00 }, + { ADAU1373_PLL_CTRL1(0), 0x00 }, + { ADAU1373_PLL_CTRL2(0), 0x00 }, + { ADAU1373_PLL_CTRL3(0), 0x00 }, + { ADAU1373_PLL_CTRL4(0), 0x00 }, + { ADAU1373_PLL_CTRL5(0), 0x00 }, + { ADAU1373_PLL_CTRL6(0), 0x02 }, + { ADAU1373_DPLL_CTRL(1), 0x00 }, + { ADAU1373_PLL_CTRL1(1), 0x00 }, + { ADAU1373_PLL_CTRL2(1), 0x00 }, + { ADAU1373_PLL_CTRL3(1), 0x00 }, + { ADAU1373_PLL_CTRL4(1), 0x00 }, + { ADAU1373_PLL_CTRL5(1), 0x00 }, + { ADAU1373_PLL_CTRL6(1), 0x02 }, + { ADAU1373_HEADDECT, 0x00 }, + { ADAU1373_ADC_CTRL, 0x00 }, + { ADAU1373_CLK_SRC_DIV(0), 0x00 }, + { ADAU1373_CLK_SRC_DIV(1), 0x00 }, + { ADAU1373_DAI(0), 0x0a }, + { ADAU1373_DAI(1), 0x0a }, + { ADAU1373_DAI(2), 0x0a }, + { ADAU1373_BCLKDIV(0), 0x00 }, + { ADAU1373_BCLKDIV(1), 0x00 }, + { ADAU1373_BCLKDIV(2), 0x00 }, + { ADAU1373_SRC_RATIOA(0), 0x00 }, + { ADAU1373_SRC_RATIOB(0), 0x00 }, + { ADAU1373_SRC_RATIOA(1), 0x00 }, + { ADAU1373_SRC_RATIOB(1), 0x00 }, + { ADAU1373_SRC_RATIOA(2), 0x00 }, + { ADAU1373_SRC_RATIOB(2), 0x00 }, + { ADAU1373_DEEMP_CTRL, 0x00 }, + { ADAU1373_SRC_DAI_CTRL(0), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(1), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(2), 0x08 }, + { ADAU1373_DIN_MIX_CTRL(0), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(1), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(2), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(3), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(4), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(0), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(1), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(2), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(3), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(4), 0x00 }, + { ADAU1373_DAI_PBL_VOL(0), 0x00 }, + { ADAU1373_DAI_PBR_VOL(0), 0x00 }, + { ADAU1373_DAI_PBL_VOL(1), 0x00 }, + { ADAU1373_DAI_PBR_VOL(1), 0x00 }, + { ADAU1373_DAI_PBL_VOL(2), 0x00 }, + { ADAU1373_DAI_PBR_VOL(2), 0x00 }, + { ADAU1373_DAI_RECL_VOL(0), 0x00 }, + { ADAU1373_DAI_RECR_VOL(0), 0x00 }, + { ADAU1373_DAI_RECL_VOL(1), 0x00 }, + { ADAU1373_DAI_RECR_VOL(1), 0x00 }, + { ADAU1373_DAI_RECL_VOL(2), 0x00 }, + { ADAU1373_DAI_RECR_VOL(2), 0x00 }, + { ADAU1373_DAC1_PBL_VOL, 0x00 }, + { ADAU1373_DAC1_PBR_VOL, 0x00 }, + { ADAU1373_DAC2_PBL_VOL, 0x00 }, + { ADAU1373_DAC2_PBR_VOL, 0x00 }, + { ADAU1373_ADC_RECL_VOL, 0x00 }, + { ADAU1373_ADC_RECR_VOL, 0x00 }, + { ADAU1373_DMIC_RECL_VOL, 0x00 }, + { ADAU1373_DMIC_RECR_VOL, 0x00 }, + { ADAU1373_VOL_GAIN1, 0x00 }, + { ADAU1373_VOL_GAIN2, 0x00 }, + { ADAU1373_VOL_GAIN3, 0x00 }, + { ADAU1373_HPF_CTRL, 0x00 }, + { ADAU1373_BASS1, 0x00 }, + { ADAU1373_BASS2, 0x00 }, + { ADAU1373_DRC(0) + 0x0, 0x78 }, + { ADAU1373_DRC(0) + 0x1, 0x18 }, + { ADAU1373_DRC(0) + 0x2, 0x00 }, + { ADAU1373_DRC(0) + 0x3, 0x00 }, + { ADAU1373_DRC(0) + 0x4, 0x00 }, + { ADAU1373_DRC(0) + 0x5, 0xc0 }, + { ADAU1373_DRC(0) + 0x6, 0x00 }, + { ADAU1373_DRC(0) + 0x7, 0x00 }, + { ADAU1373_DRC(0) + 0x8, 0x00 }, + { ADAU1373_DRC(0) + 0x9, 0xc0 }, + { ADAU1373_DRC(0) + 0xa, 0x88 }, + { ADAU1373_DRC(0) + 0xb, 0x7a }, + { ADAU1373_DRC(0) + 0xc, 0xdf }, + { ADAU1373_DRC(0) + 0xd, 0x20 }, + { ADAU1373_DRC(0) + 0xe, 0x00 }, + { ADAU1373_DRC(0) + 0xf, 0x00 }, + { ADAU1373_DRC(1) + 0x0, 0x78 }, + { ADAU1373_DRC(1) + 0x1, 0x18 }, + { ADAU1373_DRC(1) + 0x2, 0x00 }, + { ADAU1373_DRC(1) + 0x3, 0x00 }, + { ADAU1373_DRC(1) + 0x4, 0x00 }, + { ADAU1373_DRC(1) + 0x5, 0xc0 }, + { ADAU1373_DRC(1) + 0x6, 0x00 }, + { ADAU1373_DRC(1) + 0x7, 0x00 }, + { ADAU1373_DRC(1) + 0x8, 0x00 }, + { ADAU1373_DRC(1) + 0x9, 0xc0 }, + { ADAU1373_DRC(1) + 0xa, 0x88 }, + { ADAU1373_DRC(1) + 0xb, 0x7a }, + { ADAU1373_DRC(1) + 0xc, 0xdf }, + { ADAU1373_DRC(1) + 0xd, 0x20 }, + { ADAU1373_DRC(1) + 0xe, 0x00 }, + { ADAU1373_DRC(1) + 0xf, 0x00 }, + { ADAU1373_DRC(2) + 0x0, 0x78 }, + { ADAU1373_DRC(2) + 0x1, 0x18 }, + { ADAU1373_DRC(2) + 0x2, 0x00 }, + { ADAU1373_DRC(2) + 0x3, 0x00 }, + { ADAU1373_DRC(2) + 0x4, 0x00 }, + { ADAU1373_DRC(2) + 0x5, 0xc0 }, + { ADAU1373_DRC(2) + 0x6, 0x00 }, + { ADAU1373_DRC(2) + 0x7, 0x00 }, + { ADAU1373_DRC(2) + 0x8, 0x00 }, + { ADAU1373_DRC(2) + 0x9, 0xc0 }, + { ADAU1373_DRC(2) + 0xa, 0x88 }, + { ADAU1373_DRC(2) + 0xb, 0x7a }, + { ADAU1373_DRC(2) + 0xc, 0xdf }, + { ADAU1373_DRC(2) + 0xd, 0x20 }, + { ADAU1373_DRC(2) + 0xe, 0x00 }, + { ADAU1373_DRC(2) + 0xf, 0x00 }, + { ADAU1373_3D_CTRL1, 0x00 }, + { ADAU1373_3D_CTRL2, 0x00 }, + { ADAU1373_FDSP_SEL1, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL4, 0x00 }, + { ADAU1373_DIGMICCTRL, 0x00 }, + { ADAU1373_DIGEN, 0x00 }, }; static const unsigned int adau1373_out_tlv[] = { @@ -418,6 +554,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int pll_id = w->name[3] - '1'; unsigned int val; @@ -426,7 +563,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, else val = 0; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_PLL_EN, val); if (SND_SOC_DAPM_EVENT_ON(event)) @@ -938,7 +1075,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); - snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, (div << 2) | ADAU1373_BCLKDIV_64); @@ -959,7 +1096,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + return regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ADAU1373_DAI_WLEN_MASK, ctrl); } @@ -1016,7 +1153,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ~ADAU1373_DAI_WLEN_MASK, ctrl); return 0; @@ -1039,7 +1176,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, adau1373_dai->sysclk = freq; adau1373_dai->clk_src = clk_id; - snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SOURCE, clk_id << 5); return 0; @@ -1120,6 +1257,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = { static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dpll_div = 0; unsigned int x, r, n, m, i, j, mode; @@ -1187,36 +1325,36 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, if (dpll_div) { dpll_div = 11 - dpll_div; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); } else { - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, ADAU1373_PLL_CTRL6_DPLL_BYPASS); } - snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id), (source << 4) | dpll_div); - snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), (r << 3) | (x << 1) | mode); /* Set sysclk to pll_rate / 4 */ - snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); return 0; } -static void adau1373_load_drc_settings(struct snd_soc_codec *codec, +static void adau1373_load_drc_settings(struct adau1373 *adau1373, unsigned int nr, uint8_t *drc) { unsigned int i; for (i = 0; i < ADAU1373_DRC_SIZE; ++i) - snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); + regmap_write(adau1373->regmap, ADAU1373_DRC(nr) + i, drc[i]); } static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) @@ -1235,13 +1373,14 @@ static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) static int adau1373_probe(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -1256,7 +1395,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) return -EINVAL; for (i = 0; i < pdata->num_drc; ++i) { - adau1373_load_drc_settings(codec, i, + adau1373_load_drc_settings(adau1373, i, pdata->drc_setting[i]); } @@ -1268,18 +1407,18 @@ static int adau1373_probe(struct snd_soc_codec *codec) if (pdata->input_differential[i]) val |= BIT(i); } - snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + regmap_write(adau1373->regmap, ADAU1373_INPUT_MODE, val); val = 0; if (pdata->lineout_differential) val |= ADAU1373_OUTPUT_CTRL_LDIFF; if (pdata->lineout_ground_sense) val |= ADAU1373_OUTPUT_CTRL_LNFBEN; - snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + regmap_write(adau1373->regmap, ADAU1373_OUTPUT_CTRL, val); lineout_differential = pdata->lineout_differential; - snd_soc_write(codec, ADAU1373_EP_CTRL, + regmap_write(adau1373->regmap, ADAU1373_EP_CTRL, (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); } @@ -1289,7 +1428,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) ARRAY_SIZE(adau1373_lineout2_controls)); } - snd_soc_write(codec, ADAU1373_ADC_CTRL, + regmap_write(adau1373->regmap, ADAU1373_ADC_CTRL, ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); return 0; @@ -1298,17 +1437,19 @@ static int adau1373_probe(struct snd_soc_codec *codec) static int adau1373_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, 0); break; } @@ -1324,17 +1465,49 @@ static int adau1373_remove(struct snd_soc_codec *codec) static int adau1373_suspend(struct snd_soc_codec *codec) { - return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adau1373->regmap, true); + + return ret; } static int adau1373_resume(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adau1373->regmap, false); adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_cache_sync(codec); + regcache_sync(adau1373->regmap); return 0; } +static bool adau1373_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1373_SOFT_RESET: + case ADAU1373_ADC_DAC_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config adau1373_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .volatile_reg = adau1373_register_volatile, + .max_register = ADAU1373_SOFT_RESET, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1373_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults), +}; + static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .remove = adau1373_remove, @@ -1342,9 +1515,6 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), - .reg_cache_default = adau1373_default_regs, - .reg_word_size = sizeof(uint8_t), .set_pll = adau1373_set_pll, @@ -1366,6 +1536,11 @@ static int adau1373_i2c_probe(struct i2c_client *client, if (!adau1373) return -ENOMEM; + adau1373->regmap = devm_regmap_init_i2c(client, + &adau1373_regmap_config); + if (IS_ERR(adau1373->regmap)) + return PTR_ERR(adau1373->regmap); + dev_set_drvdata(&client->dev, adau1373); ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, -- cgit From 6fb04138a3068609fa0ef3a98b60e31e686b3160 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 13:47:09 +0200 Subject: ASoC: adau1373: Remove ADAU1373_PLL_CTRL7 register definition There is no such register. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index c57c1f81a611..2f84054c9b7d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -74,7 +74,6 @@ struct adau1373 { #define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7) #define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7) #define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7) -#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7) #define ADAU1373_HEADDECT 0x36 #define ADAU1373_ADC_DAC_STATUS 0x37 #define ADAU1373_ADC_CTRL 0x3c -- cgit From 729485f6adbf1c7e1e08a01d2c276da30a91b0a4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 13:47:10 +0200 Subject: ASoC: adau1373: Issue soft reset on probe Reset the device on probe to make sure that the register settings match the register cache defaults. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 2f84054c9b7d..59654b1e7f3f 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1540,6 +1540,8 @@ static int adau1373_i2c_probe(struct i2c_client *client, if (IS_ERR(adau1373->regmap)) return PTR_ERR(adau1373->regmap); + regmap_write(adau1373->regmap, ADAU1373_SOFT_RESET, 0x00); + dev_set_drvdata(&client->dev, adau1373); ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, -- cgit From 2560b3d1bdf1344aa65bba1523a08e4db27a3c14 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 15:18:25 +0200 Subject: ASoC: adav80x: Convert to direct regmap usage Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 147 +++++++++++++++++++++++++++++++-------------- 1 file changed, 102 insertions(+), 45 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 15b012d0f226..14a7c169d004 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -115,22 +115,34 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static u8 adav80x_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, - 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, - 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, - 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, - 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, - 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, - 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +static struct reg_default adav80x_reg_defaults[] = { + { ADAV80X_PLAYBACK_CTRL, 0x01 }, + { ADAV80X_AUX_IN_CTRL, 0x01 }, + { ADAV80X_REC_CTRL, 0x02 }, + { ADAV80X_AUX_OUT_CTRL, 0x01 }, + { ADAV80X_DPATH_CTRL1, 0xc0 }, + { ADAV80X_DPATH_CTRL2, 0x11 }, + { ADAV80X_DAC_CTRL1, 0x00 }, + { ADAV80X_DAC_CTRL2, 0x00 }, + { ADAV80X_DAC_CTRL3, 0x00 }, + { ADAV80X_DAC_L_VOL, 0xff }, + { ADAV80X_DAC_R_VOL, 0xff }, + { ADAV80X_PGA_L_VOL, 0x00 }, + { ADAV80X_PGA_R_VOL, 0x00 }, + { ADAV80X_ADC_CTRL1, 0x00 }, + { ADAV80X_ADC_CTRL2, 0x00 }, + { ADAV80X_ADC_L_VOL, 0xff }, + { ADAV80X_ADC_R_VOL, 0xff }, + { ADAV80X_PLL_CTRL1, 0x00 }, + { ADAV80X_PLL_CTRL2, 0x00 }, + { ADAV80X_ICLK_CTRL1, 0x00 }, + { ADAV80X_ICLK_CTRL2, 0x00 }, + { ADAV80X_PLL_CLK_SRC, 0x00 }, + { ADAV80X_PLL_OUTE, 0x00 }, }; struct adav80x { - enum snd_soc_control_type control_type; + struct regmap *regmap; enum adav80x_clk_src clk_src; unsigned int sysclk; @@ -298,7 +310,7 @@ static int adav80x_set_deemph(struct snd_soc_codec *codec) val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; } - return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + return regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); } @@ -394,10 +406,11 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, capture); - snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + regmap_write(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], + playback); adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -407,6 +420,7 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int adav80x_set_adc_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -414,7 +428,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, else val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; - snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + regmap_update_bits(adav80x->regmap, ADAV80X_ADC_CTRL1, ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); return 0; @@ -423,6 +437,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, static int adav80x_set_dac_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -430,7 +445,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, else val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, val); @@ -440,6 +455,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, struct snd_soc_dai *dai, snd_pcm_format_t format) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; switch (format) { @@ -459,7 +475,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_WORD_LEN_MASK, val); return 0; @@ -491,7 +507,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], ADAV80X_PLAYBACK_MODE_MASK, val); return 0; @@ -554,8 +570,10 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); - snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); - snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL1, + iclk_ctrl1); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2, + iclk_ctrl2); snd_soc_dapm_sync(&codec->dapm); } @@ -575,10 +593,12 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id); if (freq == 0) { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, mask); adav80x->sysclk_pd[clk_id] = true; } else { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, 0); adav80x->sysclk_pd[clk_id] = false; } @@ -650,9 +670,9 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, - pll_ctrl1); - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL1, + ADAV80X_PLL_CTRL1_PLLDIV, pll_ctrl1); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL2, ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); if (source != adav80x->pll_src) { @@ -661,7 +681,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, else pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); - snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CLK_SRC, ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); adav80x->pll_src = source; @@ -675,6 +695,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, static int adav80x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int mask = ADAV80X_DAC_CTRL1_PD; switch (level) { @@ -683,10 +704,12 @@ static int adav80x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + 0x00); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + mask); break; } @@ -780,7 +803,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -791,23 +814,31 @@ static int adav80x_probe(struct snd_soc_codec *codec) snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); /* Power down S/PDIF receiver, since it is currently not supported */ - snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20); + regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20); /* Disable DAC zero flag */ - snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int adav80x_suspend(struct snd_soc_codec *codec) { - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adav80x->regmap, true); + + return ret; } static int adav80x_resume(struct snd_soc_codec *codec) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adav80x->regmap, false); adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->cache_sync = 1; - snd_soc_cache_sync(codec); + regcache_sync(adav80x->regmap); return 0; } @@ -827,10 +858,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, - .reg_word_size = sizeof(u8), - .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), - .reg_cache_default = adav80x_default_regs, - .controls = adav80x_controls, .num_controls = ARRAY_SIZE(adav80x_controls), .dapm_widgets = adav80x_dapm_widgets, @@ -839,18 +866,21 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, - enum snd_soc_control_type control_type) +static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) { struct adav80x *adav80x; int ret; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); if (!adav80x) return -ENOMEM; + dev_set_drvdata(dev, adav80x); - adav80x->control_type = control_type; + adav80x->regmap = regmap; ret = snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); @@ -868,6 +898,19 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct regmap_config adav80x_spi_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + .read_flag_mask = 0x01, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct spi_device_id adav80x_spi_id[] = { { "adav801", 0 }, { } @@ -876,7 +919,8 @@ MODULE_DEVICE_TABLE(spi, adav80x_spi_id); static int adav80x_spi_probe(struct spi_device *spi) { - return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); + return adav80x_bus_probe(&spi->dev, + devm_regmap_init_spi(spi, &adav80x_spi_regmap_config)); } static int adav80x_spi_remove(struct spi_device *spi) @@ -896,6 +940,18 @@ static struct spi_driver adav80x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct regmap_config adav80x_i2c_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } @@ -905,7 +961,8 @@ MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - return adav80x_bus_probe(&client->dev, SND_SOC_I2C); + return adav80x_bus_probe(&client->dev, + devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config)); } static int adav80x_i2c_remove(struct i2c_client *client) -- cgit From d2724de1874917be2a12939657820616742bd8ec Mon Sep 17 00:00:00 2001 From: Hannes Gräuler Date: Sat, 28 Sep 2013 21:51:08 +0200 Subject: ALSA: snd-usb-caiaq: LED support for Maschine Controller MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds LED support for the Native Instruments Maschine Controller. It adds ALSA controls for dimming the LEDs of all buttons and the backlight of the two displays. Signed-off-by: Hannes Gräuler Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/control.c | 92 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/caiaq/device.c | 25 +++++++++++++ sound/usb/caiaq/device.h | 5 +++ 3 files changed, 122 insertions(+) diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index ae6b50f9ed56..f65fc0987cfb 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -28,6 +28,7 @@ #include "control.h" #define CNT_INTVAL 0x10000 +#define MASCHINE_BANK_SIZE 32 static int control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -105,6 +106,10 @@ static int control_put(struct snd_kcontrol *kcontrol, USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) cmd = EP1_CMD_DIMM_LEDS; + if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + cmd = EP1_CMD_DIMM_LEDS; + if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; @@ -121,6 +126,20 @@ static int control_put(struct snd_kcontrol *kcontrol, usb_sndbulkpipe(cdev->chip.dev, 8), cdev->ep8_out_buf, sizeof(cdev->ep8_out_buf), &actual_len, 200); + } else if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) { + + int bank = 0; + int offset = 0; + + if (i >= MASCHINE_BANK_SIZE) { + bank = 0x1e; + offset = MASCHINE_BANK_SIZE; + } + + snd_usb_caiaq_send_command_bank(cdev, cmd, bank, + cdev->control_state + offset, + MASCHINE_BANK_SIZE); } else { snd_usb_caiaq_send_command(cdev, cmd, cdev->control_state, sizeof(cdev->control_state)); @@ -490,6 +509,74 @@ static struct caiaq_controller kontrols4_controller[] = { { "LED: FX2: Mode", 133 | CNT_INTVAL }, }; +static struct caiaq_controller maschine_controller[] = { + { "LED: Pad 1", 3 | CNT_INTVAL }, + { "LED: Pad 2", 2 | CNT_INTVAL }, + { "LED: Pad 3", 1 | CNT_INTVAL }, + { "LED: Pad 4", 0 | CNT_INTVAL }, + { "LED: Pad 5", 7 | CNT_INTVAL }, + { "LED: Pad 6", 6 | CNT_INTVAL }, + { "LED: Pad 7", 5 | CNT_INTVAL }, + { "LED: Pad 8", 4 | CNT_INTVAL }, + { "LED: Pad 9", 11 | CNT_INTVAL }, + { "LED: Pad 10", 10 | CNT_INTVAL }, + { "LED: Pad 11", 9 | CNT_INTVAL }, + { "LED: Pad 12", 8 | CNT_INTVAL }, + { "LED: Pad 13", 15 | CNT_INTVAL }, + { "LED: Pad 14", 14 | CNT_INTVAL }, + { "LED: Pad 15", 13 | CNT_INTVAL }, + { "LED: Pad 16", 12 | CNT_INTVAL }, + + { "LED: Mute", 16 | CNT_INTVAL }, + { "LED: Solo", 17 | CNT_INTVAL }, + { "LED: Select", 18 | CNT_INTVAL }, + { "LED: Duplicate", 19 | CNT_INTVAL }, + { "LED: Navigate", 20 | CNT_INTVAL }, + { "LED: Pad Mode", 21 | CNT_INTVAL }, + { "LED: Pattern", 22 | CNT_INTVAL }, + { "LED: Scene", 23 | CNT_INTVAL }, + + { "LED: Shift", 24 | CNT_INTVAL }, + { "LED: Erase", 25 | CNT_INTVAL }, + { "LED: Grid", 26 | CNT_INTVAL }, + { "LED: Right Bottom", 27 | CNT_INTVAL }, + { "LED: Rec", 28 | CNT_INTVAL }, + { "LED: Play", 29 | CNT_INTVAL }, + { "LED: Left Bottom", 32 | CNT_INTVAL }, + { "LED: Restart", 33 | CNT_INTVAL }, + + { "LED: Group A", 41 | CNT_INTVAL }, + { "LED: Group B", 40 | CNT_INTVAL }, + { "LED: Group C", 37 | CNT_INTVAL }, + { "LED: Group D", 36 | CNT_INTVAL }, + { "LED: Group E", 39 | CNT_INTVAL }, + { "LED: Group F", 38 | CNT_INTVAL }, + { "LED: Group G", 35 | CNT_INTVAL }, + { "LED: Group H", 34 | CNT_INTVAL }, + + { "LED: Auto Write", 42 | CNT_INTVAL }, + { "LED: Snap", 43 | CNT_INTVAL }, + { "LED: Right Top", 44 | CNT_INTVAL }, + { "LED: Left Top", 45 | CNT_INTVAL }, + { "LED: Sampling", 46 | CNT_INTVAL }, + { "LED: Browse", 47 | CNT_INTVAL }, + { "LED: Step", 48 | CNT_INTVAL }, + { "LED: Control", 49 | CNT_INTVAL }, + + { "LED: Top Button 1", 57 | CNT_INTVAL }, + { "LED: Top Button 2", 56 | CNT_INTVAL }, + { "LED: Top Button 3", 55 | CNT_INTVAL }, + { "LED: Top Button 4", 54 | CNT_INTVAL }, + { "LED: Top Button 5", 53 | CNT_INTVAL }, + { "LED: Top Button 6", 52 | CNT_INTVAL }, + { "LED: Top Button 7", 51 | CNT_INTVAL }, + { "LED: Top Button 8", 50 | CNT_INTVAL }, + + { "LED: Note Repeat", 58 | CNT_INTVAL }, + + { "Backlight Display", 59 | CNT_INTVAL } +}; + static int add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *cdev) { @@ -553,6 +640,11 @@ int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *cdev) ret = add_controls(kontrols4_controller, ARRAY_SIZE(kontrols4_controller), cdev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + ret = add_controls(maschine_controller, + ARRAY_SIZE(maschine_controller), cdev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 1a61dd12fe38..bc55f708a696 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -235,6 +235,31 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, cdev->ep1_out_buf, len+1, &actual_len, 200); } +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len) +{ + int actual_len; + struct usb_device *usb_dev = cdev->chip.dev; + + if (!usb_dev) + return -EIO; + + if (len > EP1_BUFSIZE - 2) + len = EP1_BUFSIZE - 2; + + if (buffer && len > 0) + memcpy(cdev->ep1_out_buf+2, buffer, len); + + cdev->ep1_out_buf[0] = command; + cdev->ep1_out_buf[1] = bank; + + return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), + cdev->ep1_out_buf, len+2, &actual_len, 200); +} + int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bpp) { diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ad102fac6942..ab0f7520a99b 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -128,5 +128,10 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, unsigned char command, const unsigned char *buffer, int len); +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len); #endif /* CAIAQ_DEVICE_H */ -- cgit From 7cc302d231aae87a08909ae40cdf36dfe7bb5102 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 30 Sep 2013 17:08:15 +0300 Subject: ASoC: pcm: Remove extra spaces from dev_ prints dev_ prints are already prefixed by ": " before format string so there is no need for extra spaces. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6b5cb5..d4498723b375 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -721,7 +721,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); @@ -749,7 +749,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, if (dpcm->fe == fe) continue; - dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + dev_dbg(fe->dev, "reparent %s path %s %s %s\n", stream ? "capture" : "playback", dpcm->fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -773,7 +773,7 @@ static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) continue; - dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -2116,7 +2116,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- cgit From 4fa8dbc18e8a57ea21c63103abdea042ab923202 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 30 Sep 2013 22:17:37 -0300 Subject: ASoC: fsl: imx-sgtl5000: Add .remove back Commit e8f00c1b01 (Merge remote-tracking branch 'asoc/fix/fsl' into asoc-devm) fixed a conflict, but missed to add the .remove function back,which causes the following build warning: sound/soc/fsl/imx-sgtl5000.c:185:12: warning: 'imx_sgtl5000_remove' defined but not used [-Wunused-function] Fix the warning by adding the .remove function back. Reported-by: Olof Johansson Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 6f4bdc89ae3c..ed6ba1eba557 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -204,6 +204,7 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, + .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); -- cgit From e2c9917bfa4e104ba53819b37498eb45fbcd4e41 Mon Sep 17 00:00:00 2001 From: Vladimir Murzin Date: Sun, 29 Sep 2013 16:00:13 +0200 Subject: ASoC: kirkwood: fix compilation warning in kirkwood_dma_open writel() supposes the first argument of type unsigned int. This fix the warning: sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_open': sound/soc/kirkwood/kirkwood-dma.c:164:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] Signed-off-by: Vladimir Murzin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 0c85c4e1a1ae..55d0d9d3a9fd 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -159,7 +159,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) * Enable Error interrupts. We're only ack'ing them but * it's useful for diagnostics */ - writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } dram = mv_mbus_dram_info(); -- cgit From 648c538204c23370c734d72921155cc24aff928d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 1 Oct 2013 14:48:24 +0200 Subject: ASoC: tas5086: move two variables into private struct We need to access the charge_period and start_mid_z values from other places later, so move them to the private struct. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88f7204..31b5868ef7c1 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -244,6 +244,8 @@ struct tas5086_private { unsigned int mclk, sclk; unsigned int format; bool deemph; + unsigned int charge_period; + unsigned int pwm_start_mid_z; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -720,13 +722,15 @@ static const int tas5086_charge_period[] = { static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - int charge_period = 1300000; /* hardware default is 1300 ms */ - u8 pwm_start_mid_z = 0; int i, ret; + priv->pwm_start_mid_z = 0; + priv->charge_period = 1300000; /* hardware default is 1300 ms */ + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; - of_property_read_u32(of_node, "ti,charge-period", &charge_period); + of_property_read_u32(of_node, "ti,charge-period", + &priv->charge_period); for (i = 0; i < 6; i++) { char name[25]; @@ -735,7 +739,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) "ti,mid-z-channel-%d", i + 1); if (of_get_property(of_node, name, NULL) != NULL) - pwm_start_mid_z |= 1 << i; + priv->pwm_start_mid_z |= 1 << i; } } @@ -744,25 +748,25 @@ static int tas5086_probe(struct snd_soc_codec *codec) * configure 'part 1' of the PWM starts to use Mid-Z, and tell * all configured mid-z channels to start start under 'part 1'. */ - if (pwm_start_mid_z) + if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, TAS5086_PWM_START_MIDZ_FOR_START_1 | - pwm_start_mid_z); + priv->pwm_start_mid_z); /* lookup and set split-capacitor charge period */ - if (charge_period == 0) { + if (priv->charge_period == 0) { regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); } else { i = index_in_array(tas5086_charge_period, ARRAY_SIZE(tas5086_charge_period), - charge_period); + priv->charge_period); if (i >= 0) regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, i + 0x08); else dev_warn(codec->dev, "Invalid split-cap charge period of %d ns.\n", - charge_period); + priv->charge_period); } /* enable factory trim */ -- cgit From d5fd3ccc2d9df493ad6f1eaf7aba72f690e98937 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 1 Oct 2013 14:48:25 +0200 Subject: ASoC: tas5086: move initialization code to own functions We'll need to call code to initialize and reset the codec again at resume time, so factor it out first. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 128 +++++++++++++++++++++++++-------------------- 1 file changed, 72 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 31b5868ef7c1..3a88c68145c2 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -458,6 +458,75 @@ static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream) return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val); } +static void tas5086_reset(struct tas5086_private *priv) +{ + if (gpio_is_valid(priv->gpio_nreset)) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(priv->gpio_nreset, 0); + udelay(1); + gpio_set_value(priv->gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } +} + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_init(struct device *dev, struct tas5086_private *priv) +{ + int ret, i; + + /* + * If any of the channels is configured to start in Mid-Z mode, + * configure 'part 1' of the PWM starts to use Mid-Z, and tell + * all configured mid-z channels to start start under 'part 1'. + */ + if (priv->pwm_start_mid_z) + regmap_write(priv->regmap, TAS5086_PWM_START, + TAS5086_PWM_START_MIDZ_FOR_START_1 | + priv->pwm_start_mid_z); + + /* lookup and set split-capacitor charge period */ + if (priv->charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + priv->charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(dev, + "Invalid split-cap charge period of %d ns.\n", + priv->charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* mute all channels for now */ + ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + TAS5086_SOFT_MUTE_ALL); + if (ret < 0) + return ret; + + return 0; +} + /* TAS5086 controls */ static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); @@ -712,13 +781,6 @@ static const struct of_device_id tas5086_dt_ids[] = { MODULE_DEVICE_TABLE(of, tas5086_dt_ids); #endif -/* charge period values in microseconds */ -static const int tas5086_charge_period[] = { - 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, - 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, - 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, -}; - static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); @@ -729,6 +791,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; + of_property_read_u32(of_node, "ti,charge-period", &priv->charge_period); @@ -743,39 +806,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) } } - /* - * If any of the channels is configured to start in Mid-Z mode, - * configure 'part 1' of the PWM starts to use Mid-Z, and tell - * all configured mid-z channels to start start under 'part 1'. - */ - if (priv->pwm_start_mid_z) - regmap_write(priv->regmap, TAS5086_PWM_START, - TAS5086_PWM_START_MIDZ_FOR_START_1 | - priv->pwm_start_mid_z); - - /* lookup and set split-capacitor charge period */ - if (priv->charge_period == 0) { - regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); - } else { - i = index_in_array(tas5086_charge_period, - ARRAY_SIZE(tas5086_charge_period), - priv->charge_period); - if (i >= 0) - regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, - i + 0x08); - else - dev_warn(codec->dev, - "Invalid split-cap charge period of %d ns.\n", - priv->charge_period); - } - - /* enable factory trim */ - ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); - if (ret < 0) - return ret; - - /* start all channels */ - ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + ret = tas5086_init(codec->dev, priv); if (ret < 0) return ret; @@ -784,12 +815,6 @@ static int tas5086_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - /* mute all channels for now */ - ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, - TAS5086_SOFT_MUTE_ALL); - if (ret < 0) - return ret; - return 0; } @@ -866,17 +891,8 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset")) gpio_nreset = -EINVAL; - if (gpio_is_valid(gpio_nreset)) { - /* Reset codec - minimum assertion time is 400ns */ - gpio_direction_output(gpio_nreset, 0); - udelay(1); - gpio_set_value(gpio_nreset, 1); - - /* Codec needs ~15ms to wake up */ - msleep(15); - } - priv->gpio_nreset = gpio_nreset; + tas5086_reset(priv); /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); -- cgit From 25c84cc1ace56421fa9a676a387a1919e7bc4e62 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 1 Oct 2013 14:48:26 +0200 Subject: ASoC: tas5086: add suspend callback When going to suspend, shut down all channels and re-do the init procedure at resume time. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 3a88c68145c2..2996d2ea026b 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -762,14 +762,39 @@ static struct snd_soc_dai_driver tas5086_dai = { }; #ifdef CONFIG_PM +static int tas5086_soc_suspend(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + /* Shut down all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x60); + if (ret < 0) + return ret; + + return 0; +} + static int tas5086_soc_resume(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; - /* Restore codec state */ - return regcache_sync(priv->regmap); + tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + + ret = tas5086_init(codec->dev, priv); + if (ret < 0) + return ret; + + ret = regcache_sync(priv->regmap); + if (ret < 0) + return ret; + + return 0; } #else +#define tas5086_soc_suspend NULL #define tas5086_soc_resume NULL #endif /* CONFIG_PM */ @@ -832,6 +857,7 @@ static int tas5086_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { .probe = tas5086_probe, .remove = tas5086_remove, + .suspend = tas5086_soc_suspend, .resume = tas5086_soc_resume, .controls = tas5086_controls, .num_controls = ARRAY_SIZE(tas5086_controls), -- cgit From a85e419edee73ec458354388e1ba9b8b58bdcbba Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 1 Oct 2013 14:50:02 +0200 Subject: ASoC: davinci-mcasp: add support for suspend and resume When the system returns from suspend, it looses its configuration. Most of it is restored by running a normal audio stream startup, but the DAI format is left unset as that's configured on the audio device creation. Hence, it suffices here to care for the registers which are touched by davinci_mcasp_set_dai_fmt() and restore them when the system is resumed. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 12 ++++++++++++ 2 files changed, 51 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 32ddb7fe5034..cdfe959d6062 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1251,12 +1251,51 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int davinci_mcasp_suspend(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); + a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); + a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); + a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); + a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); + a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); + a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + + return 0; +} + +static int davinci_mcasp_resume(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl); + mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir); + + return 0; +} +#endif + +SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops, + davinci_mcasp_suspend, + davinci_mcasp_resume); + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a9ac0c11da71..a2e27e1c32f3 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -43,6 +43,18 @@ struct davinci_audio_dev { /* McASP FIFO related */ u8 txnumevt; u8 rxnumevt; + +#ifdef CONFIG_PM_SLEEP + struct { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; + } context; +#endif }; #endif /* DAVINCI_MCASP_H */ -- cgit From 93721039903eab4a3d406e6fb095e2598093bc9c Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sun, 6 Oct 2013 22:31:06 +0200 Subject: ALSA: usb-audio: remove unused parameter from sync_ep_set_params Since the format is not actually used in sync_ep_set_params(), there is no need to pass it down. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 21dc6422d747..5dd51af0525a 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -714,8 +714,7 @@ out_of_memory: /* * configure a sync endpoint */ -static int sync_ep_set_params(struct snd_usb_endpoint *ep, - struct audioformat *fmt) +static int sync_ep_set_params(struct snd_usb_endpoint *ep) { int i; @@ -812,7 +811,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, buffer_periods, fmt, sync_ep); break; case SND_USB_ENDPOINT_TYPE_SYNC: - err = sync_ep_set_params(ep, fmt); + err = sync_ep_set_params(ep); break; default: err = -EINVAL; -- cgit From 26de5d0a8df0817a58422f6ad43019c47716ce1f Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sun, 6 Oct 2013 22:31:07 +0200 Subject: ALSA: usb-audio: remove deactivate_endpoints() The only call site for deactivate_endpoints() at snd_usb_hw_free(). The return value is not checked there, as it is irrelevant if it fails on hw_free. This patch moves the deactivation of the endpoints directly into snd_usb_hw_free(). Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 19 ++----------------- 1 file changed, 2 insertions(+), 17 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 19e79953f2e3..1a9a01853688 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -282,22 +282,6 @@ static void stop_endpoints(struct snd_usb_substream *subs, bool wait) } } -static int deactivate_endpoints(struct snd_usb_substream *subs) -{ - int reta, retb; - - reta = snd_usb_endpoint_deactivate(subs->sync_endpoint); - retb = snd_usb_endpoint_deactivate(subs->data_endpoint); - - if (reta < 0) - return reta; - - if (retb < 0) - return retb; - - return 0; -} - static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, unsigned int altsetting, struct usb_host_interface **alts, @@ -736,7 +720,8 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) down_read(&subs->stream->chip->shutdown_rwsem); if (!subs->stream->chip->shutdown) { stop_endpoints(subs, true); - deactivate_endpoints(subs); + snd_usb_endpoint_deactivate(subs->sync_endpoint); + snd_usb_endpoint_deactivate(subs->data_endpoint); } up_read(&subs->stream->chip->shutdown_rwsem); return snd_pcm_lib_free_vmalloc_buffer(substream); -- cgit From 239b9f7990873d851335ed23fa091ce1484a5bcc Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sun, 6 Oct 2013 22:31:09 +0200 Subject: ALSA: usb-audio: don't deactivate URBs on in-use EP If an endpoint in use, its associated URBs should not be deactivated. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 5dd51af0525a..e84732c874cb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -959,12 +959,12 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) if (!ep) return -EINVAL; - deactivate_urbs(ep, true); - wait_clear_urbs(ep); - if (ep->use_count != 0) return 0; + deactivate_urbs(ep, true); + wait_clear_urbs(ep); + clear_bit(EP_FLAG_ACTIVATED, &ep->flags); return 0; -- cgit From 9b7c552bba88748001574925b80ba520691b0e4d Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sun, 6 Oct 2013 22:31:10 +0200 Subject: ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() The return value of snd_usb_endpoint_deactivate() is not used, make the function have no return value. Update the documentation to reflect what the function is actually doing. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 15 +++++---------- sound/usb/endpoint.h | 2 +- 2 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e84732c874cb..26856601422b 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -946,28 +946,23 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep) * * @ep: the endpoint to deactivate * - * If the endpoint is not currently in use, this functions will select the - * alternate interface setting 0 for the interface of this endpoint. + * If the endpoint is not currently in use, this functions will + * deactivate its associated URBs. * * In case of any active users, this functions does nothing. - * - * Returns an error if usb_set_interface() failed, 0 in all other - * cases. */ -int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) +void snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) { if (!ep) - return -EINVAL; + return; if (ep->use_count != 0) - return 0; + return; deactivate_urbs(ep, true); wait_clear_urbs(ep); clear_bit(EP_FLAG_ACTIVATED, &ep->flags); - - return 0; } /** diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 3bd02f0d226c..1c7e8ee48abc 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -22,7 +22,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep); void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); -int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); -- cgit From 06613f547a4f62c2201f70b24807d9edd4d733b0 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sun, 6 Oct 2013 22:31:11 +0200 Subject: ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED should be cleared. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 1a9a01853688..0c1a95dd5538 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -246,6 +246,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) subs->sync_endpoint->iface, subs->sync_endpoint->alt_idx); if (err < 0) { + clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); snd_printk(KERN_ERR "%d:%d:%d: cannot set interface (%d)\n", subs->dev->devnum, -- cgit From df23a2466a4961092625b487ead60f526e401062 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sun, 6 Oct 2013 22:31:13 +0200 Subject: ALSA: usb-audio: rename alt_idx to altsetting As Clemens Ladisch kindly explained: "Please note that there are two methods to identify alternate settings: the number, which is the value in bAlternateSetting, and the index, which is the index in the descriptor array. There might be some wording in the USB spec that these two values must be the same, but in reality, [insert standard rant about firmware writers], bAlternateSetting must be treated as a random ID value." This patch changes the name to express the correct usage semantics. No functional change. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 +- sound/usb/endpoint.c | 6 +++--- sound/usb/pcm.c | 6 +++--- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index ca98a9b915c9..9867ab866857 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -96,7 +96,7 @@ struct snd_usb_endpoint { unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ unsigned char silence_value; unsigned int stride; - int iface, alt_idx; + int iface, altsetting; int skip_packets; /* quirks for devices to ignore the first n packets in a stream */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 26856601422b..e25e32a8824d 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -426,9 +426,9 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, list_for_each_entry(ep, &chip->ep_list, list) { if (ep->ep_num == ep_num && ep->iface == alts->desc.bInterfaceNumber && - ep->alt_idx == alts->desc.bAlternateSetting) { + ep->altsetting == alts->desc.bAlternateSetting) { snd_printdd(KERN_DEBUG "Re-using EP %x in iface %d,%d @%p\n", - ep_num, ep->iface, ep->alt_idx, ep); + ep_num, ep->iface, ep->altsetting, ep); goto __exit_unlock; } } @@ -447,7 +447,7 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, ep->type = type; ep->ep_num = ep_num; ep->iface = alts->desc.bInterfaceNumber; - ep->alt_idx = alts->desc.bAlternateSetting; + ep->altsetting = alts->desc.bAlternateSetting; INIT_LIST_HEAD(&ep->ready_playback_urbs); ep_num &= USB_ENDPOINT_NUMBER_MASK; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0c1a95dd5538..ca3256d6fde3 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -241,17 +241,17 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) struct snd_usb_endpoint *ep = subs->sync_endpoint; if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) { + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { err = usb_set_interface(subs->dev, subs->sync_endpoint->iface, - subs->sync_endpoint->alt_idx); + subs->sync_endpoint->altsetting); if (err < 0) { clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); snd_printk(KERN_ERR "%d:%d:%d: cannot set interface (%d)\n", subs->dev->devnum, subs->sync_endpoint->iface, - subs->sync_endpoint->alt_idx, err); + subs->sync_endpoint->altsetting, err); return -EIO; } } -- cgit From 05c79b772f0236b857abf8034d9b0af2ecae901c Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sun, 6 Oct 2013 22:31:17 +0200 Subject: ALSA: usb-audio: remove unused endpoint flag EP_FLAG_ACTIVATED EP_FLAG_ACTIVATED is never tested for, remove it. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e25e32a8824d..b9ba0fcc45df 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -33,7 +33,6 @@ #include "pcm.h" #include "quirks.h" -#define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 #define EP_FLAG_STOPPING 2 @@ -961,8 +960,6 @@ void snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) deactivate_urbs(ep, true); wait_clear_urbs(ep); - - clear_bit(EP_FLAG_ACTIVATED, &ep->flags); } /** -- cgit From da83fea6122ea637be5f960b95bb599561617319 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 5 Oct 2013 19:26:17 +0200 Subject: ASoC: dapm: Ignore VMID widgets for target bias VMID widgets behave very similar to signal generator widgets. Both are always considered to be powered up. This means that we need to ignore the VMID widgets in the same way as signal generator widgets when calculating the DAPM context's target bias level. Otherwise the presence of a VMID widget, regardless whether it is on an active path or not, will cause the DAPM context to be powered up. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c394df..177f8a1938da 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1840,6 +1840,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) */ switch (w->id) { case snd_soc_dapm_siggen: + case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: -- cgit From 56cac413dd6d43af8355f5d1f90a199b540f73fc Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sat, 5 Oct 2013 02:25:38 +0300 Subject: ALSA: hda - hdmi: Fix reported channel map on common default layouts hdmi_setup_fake_chmap() is supposed to set the reported channel map when the channel map is not specified by the user. However, the function indexes channel_allocations[] with a wrong value and extracts the wrong nibble from hdmi_channel_mapping[], causing wrong channel maps to be shown. Fix those issues. Tested on Intel HDMI to correctly generate various channel maps, for example 3,4,14,15,7,8,5,6 (instead of incorrect 3,4,8,7,5,6,14,0) for standard 7.1 channel audio. (Note that the side and rear channels are reported as RL/RR and RLC/RRC, respectively, as per the CEA-861 standard, instead of the more traditional SL/SR and RL/RR.) Note that this only fixes the layouts that only contain traditional 7.1 speakers (2.0, 2.1, 4.0, 5.1, 7.1, etc.). E.g. the rear center of 6.1 is still being shown wrongly due to an issue with from_cea_slot() which will be fixed in a later patch. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7ea0245fc6bd..dcc4c1cb0ab1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -740,9 +740,10 @@ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, static void hdmi_setup_fake_chmap(unsigned char *map, int ca) { int i; + int ordered_ca = get_channel_allocation_order(ca); for (i = 0; i < 8; i++) { - if (i < channel_allocations[ca].channels) - map[i] = from_cea_slot((hdmi_channel_mapping[ca][i] >> 4) & 0x0f); + if (i < channel_allocations[ordered_ca].channels) + map[i] = from_cea_slot(hdmi_channel_mapping[ca][i] & 0x0f); else map[i] = 0; } -- cgit From 249ce1387b7739dbea2ac1a697e4bf1e37ec06b7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Oct 2013 13:43:49 +0200 Subject: ASoC: dapm: Add support for virtual mixer controls This patch adds support for virtual DAPM mixer controls. They are similar to virtual DAPM enums. There is no hardware register backing the control, so changing the control's value wont have any direct effect on the hardware. But it still influences the DAPM graph by causing the path it sits on to be connected or disconnected. This in turn can cause power changes for some of the widgets on the DAPM graph, which will then modify the hardware state. Signed-off-by: Lars-Peter Clausen Tested-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 4 ++++ include/sound/soc.h | 3 ++- sound/soc/soc-dapm.c | 45 +++++++++++++++++++++++++++------------------ 3 files changed, 33 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 27a72d5d4b00..2037c45adfe6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -286,6 +286,8 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } +#define SOC_DAPM_SINGLE_VIRT(xname, max) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0) #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -300,6 +302,8 @@ struct device; .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..b429dba57bf6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1088,7 +1088,8 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift; + int reg, rreg; + unsigned int shift, rshift; unsigned int invert:1; unsigned int autodisable:1; }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 177f8a1938da..9273216f22fc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -499,18 +499,22 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrol_news[i].private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = soc_widget_read(w, reg); - val = (val >> shift) & mask; - if (invert) - val = max - val; + if (reg != SND_SOC_NOPM) { + val = soc_widget_read(w, reg); + val = (val >> shift) & mask; + if (invert) + val = max - val; + p->connect = !!val; + } else { + p->connect = 0; + } - p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -2792,7 +2796,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2805,7 +2809,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol)) + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) val = (snd_soc_read(codec, reg) >> shift) & mask; else val = dapm_kcontrol_get_value(kcontrol); @@ -2836,7 +2840,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2858,19 +2862,24 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm_kcontrol_set_value(kcontrol, val); + change = dapm_kcontrol_set_value(kcontrol, val); - mask = mask << shift; - val = val << shift; + if (reg != SND_SOC_NOPM) { + mask = mask << shift; + val = val << shift; + + change = snd_soc_test_bits(codec, reg, mask, val); + } - change = snd_soc_test_bits(codec, reg, mask, val); if (change) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + if (reg != SND_SOC_NOPM) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - card->update = &update; + card->update = &update; + } soc_dapm_mixer_update_power(card, kcontrol, connect); -- cgit From 90f28002110d783f49639f0db2ccdc0b58302cbd Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sat, 5 Oct 2013 02:25:39 +0300 Subject: ALSA: hda - hdmi: Fix incorrect default channel mapping for unusual CAs hdmi_std_setup_channel_mapping() selects a Channel Allocation according to the sink reported speaker mask, preferring the ALSA standard layouts. If the channel allocation is not one of the ALSA standard layouts, the ALSA channels are mapped directly to HDMI channels in order. However, the function does not take into account that there a holes in the HDMI channel map. Additionally, the function tries to disable a slot by using AC_VERB_SET_CHAN_SLOT with parameter ((alsa_ch << 8) | 0xf), while the correct parameter is ((0xf << 8) | hdmi_slot), i.e. the slot should be unassigned, not the ALSA channel. Fix both of the issues for non-ALSA-default layouts. Tested on Intel HDMI with a speaker mask of FL | FR | FC | RC, which causes CA 0x06 to be selected for 4-channel audio, which causes incorrect output (sound destined to RC goes to FC and FC goes nowhere) without the patch. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index dcc4c1cb0ab1..b187cce64fa4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -595,22 +595,32 @@ static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, bool non_pcm, int ca) { + struct cea_channel_speaker_allocation *ch_alloc; int i; int err; int order; int non_pcm_mapping[8]; order = get_channel_allocation_order(ca); + ch_alloc = &channel_allocations[order]; if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[order].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); - for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + int hdmi_slot = 0; + /* fill actual channel mappings in ALSA channel (i) order */ + for (i = 0; i < ch_alloc->channels; i++) { + while (!ch_alloc->speakers[7 - hdmi_slot] && !WARN_ON(hdmi_slot >= 8)) + hdmi_slot++; /* skip zero slots */ + + hdmi_channel_mapping[ca][i] = (i << 4) | hdmi_slot++; + } + /* fill the rest of the slots with ALSA channel 0xf */ + for (hdmi_slot = 0; hdmi_slot < 8; hdmi_slot++) + if (!ch_alloc->speakers[7 - hdmi_slot]) + hdmi_channel_mapping[ca][i++] = (0xf << 4) | hdmi_slot; } if (non_pcm) { - for (i = 0; i < channel_allocations[order].channels; i++) + for (i = 0; i < ch_alloc->channels; i++) non_pcm_mapping[i] = i | (i << 4); for (; i < 8; i++) non_pcm_mapping[i] = 0xf | (i << 4); -- cgit From 1df5a06abbaa876ecc01ea84064cdffb4f52a1a1 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sat, 5 Oct 2013 02:25:40 +0300 Subject: ALSA: hda - hdmi: Fix programmed active channel count Currently the converter channel count is set to the number of actual input channels. The audio infoframe channel count field is set similarly. However, sometimes the used channel map does not map all input channels to outputs. Notably, 3 channel modes (e.g. 2.1) require a dummy input channel so there are 4 input channels. According to the HDA specification, converter channel count should be programmed according to the number of _active_ channels. On Intel HDMI codecs (but not on NVIDIA), setting the converter channel to a higher value than there are actually mapped channels to HDMI slots will cause no audio to be output at all. Note that the effects of this issue are currently partially masked by other bugs that prevent the driver from actually unmapping channels in certain cases. For example, if a 4 channel stream is first created and prepared, it gets a FL,FR,RL,RR mapping (ALSA->HDMI slot mapping 0->0, 1->1, 2->4, 3->5). If one thereafter assigns a FR,FL,FC mapping to it, the driver will remap 2->3 but fail to unmap 2->4 and 3->5, so there are still 4 active channels and the issue will not trigger in this case. These bugs will be fixed separately. Fix the channel counts in the converter channel count field and in the audio infoframe channel count field to match the actual number of active channels. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b187cce64fa4..4efaca89d805 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -63,6 +63,7 @@ struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; + hda_nid_t cvt_nid; struct hda_codec *codec; struct hdmi_eld sink_eld; @@ -900,8 +901,9 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, { hda_nid_t pin_nid = per_pin->pin_nid; int channels = per_pin->channels; + int active_channels; struct hdmi_eld *eld; - int ca; + int ca, ordered_ca; union audio_infoframe ai; if (!channels) @@ -923,6 +925,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (ca < 0) ca = 0; + ordered_ca = get_channel_allocation_order(ca); + active_channels = channel_allocations[ordered_ca].channels; + + hdmi_set_channel_count(codec, per_pin->cvt_nid, active_channels); + memset(&ai, 0, sizeof(ai)); if (eld->info.conn_type == 0) { /* HDMI */ struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; @@ -930,7 +937,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hdmi_ai->type = 0x84; hdmi_ai->ver = 0x01; hdmi_ai->len = 0x0a; - hdmi_ai->CC02_CT47 = channels - 1; + hdmi_ai->CC02_CT47 = active_channels - 1; hdmi_ai->CA = ca; hdmi_checksum_audio_infoframe(hdmi_ai); } else if (eld->info.conn_type == 1) { /* DisplayPort */ @@ -939,7 +946,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, dp_ai->type = 0x84; dp_ai->len = 0x1b; dp_ai->ver = 0x11 << 2; - dp_ai->CC02_CT47 = channels - 1; + dp_ai->CC02_CT47 = active_channels - 1; dp_ai->CA = ca; } else { snd_printd("HDMI: unknown connection type at pin %d\n", @@ -957,7 +964,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, snd_printdd("hdmi_setup_audio_infoframe: " "pin=%d channels=%d\n", pin_nid, - channels); + active_channels); hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, channels, per_pin->chmap, per_pin->chmap_set); @@ -1230,6 +1237,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ per_cvt->assigned = 1; + per_pin->cvt_nid = per_cvt->cvt_nid; hinfo->nid = per_cvt->cvt_nid; snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, @@ -1552,8 +1560,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, per_pin->channels = substream->runtime->channels; per_pin->setup = true; - hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); -- cgit From 290c348ee5522a5682c2011fa4d51f232404e8a4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Oct 2013 13:43:51 +0200 Subject: ASoC: twl6040: Use virtual DAPM mixer controls By using the new virtual DAPM mixer controls it is possible to remove the twl6040 specific implementation of virtual controls. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 35059a242fa4..f2f4bcb2ff71 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -54,12 +54,7 @@ enum twl6040_dai_id { #define TWL6040_OUTHF_0dB 0x03 #define TWL6040_OUTHF_M52dB 0x1D -/* Shadow register used by the driver */ -#define TWL6040_REG_SW_SHADOW 0x2F -#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) - -/* TWL6040_REG_SW_SHADOW (0x2F) fields */ -#define TWL6040_EAR_PATH_ENABLE 0x01 +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) struct twl6040_jack_data { struct snd_soc_jack *jack; @@ -135,8 +130,6 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* REG_HFOTRIM 0x2C */ 0x09, /* REG_ACCCTL 0x2D */ 0x00, /* REG_STATUS 0x2E (ro) */ - - 0x00, /* REG_SW_SHADOW 0x2F - Shadow, non HW register */ }; /* List of registers to be restored after power up */ @@ -220,12 +213,8 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - if (likely(reg < TWL6040_REG_SW_SHADOW)) { - value = twl6040_reg_read(twl6040, reg); - twl6040_write_reg_cache(codec, reg, value); - } else { - value = twl6040_read_reg_cache(codec, reg); - } + value = twl6040_reg_read(twl6040, reg); + twl6040_write_reg_cache(codec, reg, value); return value; } @@ -261,8 +250,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - if (likely(reg < TWL6040_REG_SW_SHADOW) && - twl6040_is_path_unmuted(codec, reg)) + if (twl6040_is_path_unmuted(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else return 0; @@ -555,7 +543,7 @@ static const struct snd_kcontrol_new hfr_mux_controls = SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]); static const struct snd_kcontrol_new ep_path_enable_control = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); static const struct snd_kcontrol_new auxl_switch_control = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 6, 1, 0); -- cgit From 11f7c52d90b21a51b0bc6a8b642c6ed150bdc219 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sat, 5 Oct 2013 02:25:41 +0300 Subject: ALSA: hda - hdmi: Fix unused slots being enabled in manual and non-PCM mappings hdmi_manual_setup_channel_mapping() and hdmi_std_setup_channel_mapping try to assign ALSA channels to HDMI channel slots and disable (i.e. silence) other slots. However, they try to disable a slot by using AC_VERB_SET_CHAN_SLOT with parameter ((alsa_ch << 8) | 0xf), while the correct parameter is ((0xf << 8) | hdmi_slot), i.e. the slot should be unassigned, not the ALSA channel. Fix that by actually disabling the unused slots. Note that this bug did not cause any (reported) issues because slots incorrectly having audio are normally ignored by a receiver if the CEA channel allocation used does not map that slot to any speaker. Additionally, the converter channel count configuration limits the number of actually active channels in any case. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 29 +++++++++++++++++++---------- 1 file changed, 19 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4efaca89d805..8bf526ea8cf2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -622,9 +622,9 @@ static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, if (non_pcm) { for (i = 0; i < ch_alloc->channels; i++) - non_pcm_mapping[i] = i | (i << 4); + non_pcm_mapping[i] = (i << 4) | i; for (; i < 8; i++) - non_pcm_mapping[i] = 0xf | (i << 4); + non_pcm_mapping[i] = (0xf << 4) | i; } for (i = 0; i < 8; i++) { @@ -678,7 +678,7 @@ static int to_cea_slot(unsigned char c) if (t->map == c) return t->cea_slot; } - return 0x0f; + return -1; } /* from CEA slot to ALSA API channel position */ @@ -731,14 +731,23 @@ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, int chs, unsigned char *map) { - int i; - for (i = 0; i < 8; i++) { + int alsa_pos, hdmi_slot; + int assignments[8] = {[0 ... 7] = 0xf}; + + for (alsa_pos = 0; alsa_pos < chs; alsa_pos++) { + + hdmi_slot = to_cea_slot(map[alsa_pos]); + + if (hdmi_slot < 0) + continue; /* unassigned channel */ + + assignments[hdmi_slot] = alsa_pos; + } + + for (hdmi_slot = 0; hdmi_slot < 8; hdmi_slot++) { int val, err; - if (i < chs) - val = to_cea_slot(map[i]); - else - val = 0xf; - val |= (i << 4); + + val = (assignments[hdmi_slot] << 4) | hdmi_slot; err = snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, val); if (err) -- cgit From a5b7d510b2220cccbcaeb1b87a6d8c47efeb154c Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sat, 5 Oct 2013 02:25:42 +0300 Subject: ALSA: hda - hdmi: Fix channel maps with less common speakers For some speakers and slots the CEA slot <-> speaker assignment depends on the used CEA Channel Allocation value. Therefore the from_cea_slot() and to_cea_slot() helpers currently only work correctly for the regular 7.1 speakers. Fix them to work with all speakers, taking the re-ordered CA index as input and adapting use sites accordingly. This change allows manual channel mapping to actually work for all CEA allocated speakers. Additionally, this fixes incorrect channel map reporting in automatic channel mapping mode when an affected speaker position is used (e.g. 6.1 map which contains an RC speaker). Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 70 +++++++++++++++++++++++++++------------------- 1 file changed, 41 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8bf526ea8cf2..dab2af762d94 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -643,19 +643,27 @@ static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, struct channel_map_table { unsigned char map; /* ALSA API channel map position */ - unsigned char cea_slot; /* CEA slot value */ int spk_mask; /* speaker position bit mask */ }; static struct channel_map_table map_tables[] = { - { SNDRV_CHMAP_FL, 0x00, FL }, - { SNDRV_CHMAP_FR, 0x01, FR }, - { SNDRV_CHMAP_RL, 0x04, RL }, - { SNDRV_CHMAP_RR, 0x05, RR }, - { SNDRV_CHMAP_LFE, 0x02, LFE }, - { SNDRV_CHMAP_FC, 0x03, FC }, - { SNDRV_CHMAP_RLC, 0x06, RLC }, - { SNDRV_CHMAP_RRC, 0x07, RRC }, + { SNDRV_CHMAP_FL, FL }, + { SNDRV_CHMAP_FR, FR }, + { SNDRV_CHMAP_RL, RL }, + { SNDRV_CHMAP_RR, RR }, + { SNDRV_CHMAP_LFE, LFE }, + { SNDRV_CHMAP_FC, FC }, + { SNDRV_CHMAP_RLC, RLC }, + { SNDRV_CHMAP_RRC, RRC }, + { SNDRV_CHMAP_RC, RC }, + { SNDRV_CHMAP_FLC, FLC }, + { SNDRV_CHMAP_FRC, FRC }, + { SNDRV_CHMAP_FLH, FLH }, + { SNDRV_CHMAP_FRH, FRH }, + { SNDRV_CHMAP_FLW, FLW }, + { SNDRV_CHMAP_FRW, FRW }, + { SNDRV_CHMAP_TC, TC }, + { SNDRV_CHMAP_FCH, FCH }, {} /* terminator */ }; @@ -671,25 +679,19 @@ static int to_spk_mask(unsigned char c) } /* from ALSA API channel position to CEA slot */ -static int to_cea_slot(unsigned char c) +static int to_cea_slot(int ordered_ca, unsigned char pos) { - struct channel_map_table *t = map_tables; - for (; t->map; t++) { - if (t->map == c) - return t->cea_slot; - } - return -1; -} + int mask = to_spk_mask(pos); + int i; -/* from CEA slot to ALSA API channel position */ -static int from_cea_slot(unsigned char c) -{ - struct channel_map_table *t = map_tables; - for (; t->map; t++) { - if (t->cea_slot == c) - return t->map; + if (mask) { + for (i = 0; i < 8; i++) { + if (channel_allocations[ordered_ca].speakers[7 - i] == mask) + return i; + } } - return 0; + + return -1; } /* from speaker bit mask to ALSA API channel position */ @@ -703,6 +705,14 @@ static int spk_to_chmap(int spk) return 0; } +/* from CEA slot to ALSA API channel position */ +static int from_cea_slot(int ordered_ca, unsigned char slot) +{ + int mask = channel_allocations[ordered_ca].speakers[7 - slot]; + + return spk_to_chmap(mask); +} + /* get the CA index corresponding to the given ALSA API channel map */ static int hdmi_manual_channel_allocation(int chs, unsigned char *map) { @@ -729,14 +739,16 @@ static int hdmi_manual_channel_allocation(int chs, unsigned char *map) /* set up the channel slots for the given ALSA API channel map */ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, - int chs, unsigned char *map) + int chs, unsigned char *map, + int ca) { + int ordered_ca = get_channel_allocation_order(ca); int alsa_pos, hdmi_slot; int assignments[8] = {[0 ... 7] = 0xf}; for (alsa_pos = 0; alsa_pos < chs; alsa_pos++) { - hdmi_slot = to_cea_slot(map[alsa_pos]); + hdmi_slot = to_cea_slot(ordered_ca, map[alsa_pos]); if (hdmi_slot < 0) continue; /* unassigned channel */ @@ -763,7 +775,7 @@ static void hdmi_setup_fake_chmap(unsigned char *map, int ca) int ordered_ca = get_channel_allocation_order(ca); for (i = 0; i < 8; i++) { if (i < channel_allocations[ordered_ca].channels) - map[i] = from_cea_slot(hdmi_channel_mapping[ca][i] & 0x0f); + map[i] = from_cea_slot(ordered_ca, hdmi_channel_mapping[ca][i] & 0x0f); else map[i] = 0; } @@ -776,7 +788,7 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, { if (!non_pcm && chmap_set) { hdmi_manual_setup_channel_mapping(codec, pin_nid, - channels, map); + channels, map, ca); } else { hdmi_std_setup_channel_mapping(codec, pin_nid, non_pcm, ca); hdmi_setup_fake_chmap(map, ca); -- cgit From bb731f2100e614a8d7c5965d3663aed893859733 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sat, 5 Oct 2013 02:25:43 +0300 Subject: ALSA: hda - hdmi: Fix available channel maps missing from TLV Currently the available channel maps TLV only contains channel maps that are limited to the traditional 7.1 speakers. Since the other HDMI channel mapping functions have been fixed to properly handle all CEA-861-E specified speakers, allow them to be listed. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index dab2af762d94..62b1e852d002 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1658,8 +1658,6 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; - const unsigned int valid_mask = - FL | FR | RL | RR | LFE | FC | RLC | RRC; unsigned int __user *dst; int chs, count = 0; @@ -1677,8 +1675,6 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, int chs_bytes = chs * 4; if (cap->channels != chs) continue; - if (cap->spk_mask & ~valid_mask) - continue; if (size < 8) return -ENOMEM; if (put_user(SNDRV_CTL_TLVT_CHMAP_VAR, dst) || -- cgit From 980b24958f0c615fd003d37f0fce4ab1ecd01784 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sat, 5 Oct 2013 02:25:44 +0300 Subject: ALSA: hda - hdmi: Tweak debug messages to be more useful Allow channel map debugging for both automatic and manual channel maps, and print CA always when updating infoframe. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 62b1e852d002..00e041333904 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -637,8 +637,6 @@ static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, break; } } - - hdmi_debug_channel_mapping(codec, pin_nid); } struct channel_map_table { @@ -793,6 +791,8 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, hdmi_std_setup_channel_mapping(codec, pin_nid, non_pcm, ca); hdmi_setup_fake_chmap(map, ca); } + + hdmi_debug_channel_mapping(codec, pin_nid); } /* @@ -983,9 +983,9 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, sizeof(ai))) { snd_printdd("hdmi_setup_audio_infoframe: " - "pin=%d channels=%d\n", + "pin=%d channels=%d ca=0x%02x\n", pin_nid, - active_channels); + active_channels, ca); hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, channels, per_pin->chmap, per_pin->chmap_set); -- cgit From 052901f42f360062f36cc5c0aa6e5ae372fe0895 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Oct 2013 13:43:50 +0200 Subject: ASoC: twl4030: Use virtual DAPM mixer controls By using the new virtual DAPM mixer controls it is possible to remove the twl4030 specific implementation of virtual controls. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 80 +++++++++++++++++++++------------------------- 1 file changed, 36 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1e3884d6b3fb..dfc51bb425da 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -46,13 +46,7 @@ /* TWL4030 PMBR1 Register GPIO6 mux bits */ #define TWL4030_GPIO6_PWM0_MUTE(value) ((value & 0x03) << 2) -/* Shadow register used by the audio driver */ -#define TWL4030_REG_SW_SHADOW 0x4A -#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) - -/* TWL4030_REG_SW_SHADOW (0x4A) Fields */ -#define TWL4030_HFL_EN 0x01 -#define TWL4030_HFR_EN 0x02 +#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) /* * twl4030 register cache & default register settings @@ -132,7 +126,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VIBRA_PWM_SET (0x47) */ 0x00, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ - 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */ }; /* codec private data */ @@ -198,42 +191,41 @@ static int twl4030_write(struct snd_soc_codec *codec, int write_to_reg = 0; twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) { - /* Decide if the given register can be written */ - switch (reg) { - case TWL4030_REG_EAR_CTL: - if (twl4030->earpiece_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PREDL_CTL: - if (twl4030->predrivel_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PREDR_CTL: - if (twl4030->predriver_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PRECKL_CTL: - if (twl4030->carkitl_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PRECKR_CTL: - if (twl4030->carkitr_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_HS_GAIN_SET: - if (twl4030->hsl_enabled || twl4030->hsr_enabled) - write_to_reg = 1; - break; - default: - /* All other register can be written */ + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) write_to_reg = 1; - break; - } - if (write_to_reg) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - value, reg); + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + return 0; } @@ -532,7 +524,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Left virtual mute */ static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control = - SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = @@ -548,7 +540,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); /* Handsfree Right virtual mute */ static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control = - SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); /* Vibra */ /* Vibra audio path selection */ -- cgit From df3774c5c53305eda4053d5c941bb17a2145e3c6 Mon Sep 17 00:00:00 2001 From: Thomas Pugliese Date: Tue, 1 Oct 2013 14:32:15 -0500 Subject: ALSA: usb-audio: add support for wireless USB devices This patch updates snd_usb_audio_create also support devices whose speed == USB_SPEED_WIRELESS. Signed-off-by: Thomas Pugliese Acked-by: Greg Kroah-Hartman Signed-off-by: Takashi Iwai --- sound/usb/card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index d1f54dfe41d5..9d9de8d53469 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -346,6 +346,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, case USB_SPEED_LOW: case USB_SPEED_FULL: case USB_SPEED_HIGH: + case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: break; default: -- cgit From 6d5eba5aac4ec9bba3d6b3e6425d857706d12ccb Mon Sep 17 00:00:00 2001 From: Thomas Pugliese Date: Tue, 1 Oct 2013 14:32:57 -0500 Subject: ALSA: usb-audio: support wireless devices in snd_usb_parse_datainterval This patch adds support for dev speed USB_SPEED_WIRELESS in snd_usb_parse_datainterval which allows the usb sound core to create ISO urbs with the correct number and size of buffers. Signed-off-by: Thomas Pugliese Acked-by: Greg Kroah-Hartman Signed-off-by: Takashi Iwai --- sound/usb/helper.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 620902463c6e..51ed1ac825fd 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -118,6 +118,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, { switch (snd_usb_get_speed(chip->dev)) { case USB_SPEED_HIGH: + case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: if (get_endpoint(alts, 0)->bInterval >= 1 && get_endpoint(alts, 0)->bInterval <= 4) -- cgit From 6b5a7c66ce396242e504dbf3d9d5c06d8b1aa488 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Wed, 9 Oct 2013 17:22:32 +0530 Subject: ALSA: usb-audio: Use module_usb_driver module_usb_driver makes code simpler by removing the boilerplate. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/usb/card.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9d9de8d53469..d979050e6a6a 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -751,15 +751,4 @@ static struct usb_driver usb_audio_driver = { .supports_autosuspend = 1, }; -static int __init snd_usb_audio_init(void) -{ - return usb_register(&usb_audio_driver); -} - -static void __exit snd_usb_audio_cleanup(void) -{ - usb_deregister(&usb_audio_driver); -} - -module_init(snd_usb_audio_init); -module_exit(snd_usb_audio_cleanup); +module_usb_driver(usb_audio_driver); -- cgit From cd4035e814b04c009ece9939396d06370c53ed44 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 10 Oct 2013 09:01:25 +0200 Subject: ALSA: hda - Enable surround speakers (when line out is also present) In the case where we have both line out and more than stereo speakers, the speaker DACs will end up in extra_out_nid. In fact, AFAIU, speakers are the only ones that can end up in extra_out_nid, and if we have several of those, they should be surround outputs rather than copy front. BugLink: https://bugs.launchpad.net/bugs/1236965 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3c92ca..68801ba001ce 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5395,11 +5395,6 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, mout->hp_out_nid[i], stream_tag, 0, format); - for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) - if (!mout->no_share_stream && mout->extra_out_nid[i]) - snd_hda_codec_setup_stream(codec, - mout->extra_out_nid[i], - stream_tag, 0, format); /* surrounds */ for (i = 1; i < mout->num_dacs; i++) { @@ -5410,6 +5405,20 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); } + + /* extra surrounds */ + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) { + int ch = 0; + if (!mout->extra_out_nid[i]) + break; + if (chs >= (i + 1) * 2) + ch = i * 2; + else if (!mout->no_share_stream) + break; + snd_hda_codec_setup_stream(codec, mout->extra_out_nid[i], + stream_tag, ch, format); + } + return 0; } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); -- cgit From eb270e98e15b9f4303b074ba5d88ee98110bc451 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Oct 2013 13:52:52 +0100 Subject: ASoC: dapm: Use async I/O for DAPM sequences Within a DAPM sequence we normally don't care about when exactly a register write has completed so long as they happen in the order we requested. This means that we can issue most of the writes we do asynchronously which should maximise the ability of the underlying frameworks to keep the hardware busy, providing a small performance improvement on some systems. We currently ensure that all writes are completed both when changing to a different device and when calling into the regulator and clock frameworks. This should ensure that the previous ordering is maintained. We also ensure that writes are completed prior to calling into widget event functions since some event functions implement delays. This should be improved in future so that widgets can disable this sync in order to add extra writes. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9273216f22fc..1dbc5f8cdc98 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -409,6 +409,12 @@ static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) mutex_unlock(&w->platform->mutex); } +static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec && dapm->codec->using_regmap) + regmap_async_complete(dapm->codec->control_data); +} + static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { @@ -417,8 +423,9 @@ static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, int ret; if (w->codec && w->codec->using_regmap) { - ret = regmap_update_bits_check(w->codec->control_data, - reg, mask, value, &change); + ret = regmap_update_bits_check_async(w->codec->control_data, + reg, mask, value, + &change); if (ret != 0) return ret; } else { @@ -1201,6 +1208,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, { int ret; + soc_dapm_async_complete(w->dapm); + if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, false); @@ -1234,6 +1243,8 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, if (!w->clk) return -EIO; + soc_dapm_async_complete(w->dapm); + #ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); @@ -1426,6 +1437,7 @@ static void dapm_seq_check_event(struct snd_soc_card *card, if (w->event && (w->event_flags & event)) { pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n", w->name, ev_name); + soc_dapm_async_complete(w->dapm); trace_snd_soc_dapm_widget_event_start(w, event); ret = w->event(w, NULL, event); trace_snd_soc_dapm_widget_event_done(w, event); @@ -1498,6 +1510,7 @@ static void dapm_seq_run(struct snd_soc_card *card, struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; + struct snd_soc_dapm_context *d; LIST_HEAD(pending); int cur_sort = -1; int cur_subseq = -1; @@ -1528,6 +1541,9 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_subseq); } + if (cur_dapm && w->dapm != cur_dapm) + soc_dapm_async_complete(cur_dapm); + INIT_LIST_HEAD(&pending); cur_sort = -1; cur_subseq = INT_MIN; @@ -1586,6 +1602,10 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_dapm->seq_notifier(cur_dapm, i, cur_subseq); } + + list_for_each_entry(d, &card->dapm_list, list) { + soc_dapm_async_complete(d); + } } static void dapm_widget_update(struct snd_soc_card *card) -- cgit From 1dd275b60e5db4d0bb3763490b519176dcfc4308 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Oct 2013 13:56:37 +0100 Subject: ASoC: dapm: Run clock and regulator events separately to other supplies In order to avoid trying to use an external clock or supply for an on-chip supply prior to it being enabled move the clock and regulator supply events to a separate step in DAPM sequencing from normal supply events. This should have minimal practical impact since these widgets are sorted using SND_SOC_NOPM which is a negative value and hence sorted separately to any real register writes, though it may be relevant if supplies have event callbacks only. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 50 +++++++++++++++++++++++++------------------------- 1 file changed, 25 insertions(+), 25 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1dbc5f8cdc98..2fb0b72d8a3c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -59,31 +59,31 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_clock_supply] = 1, - [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_supply] = 2, + [snd_soc_dapm_micbias] = 3, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai_in] = 3, - [snd_soc_dapm_dai_out] = 3, - [snd_soc_dapm_aif_in] = 3, - [snd_soc_dapm_aif_out] = 3, - [snd_soc_dapm_mic] = 4, - [snd_soc_dapm_mux] = 5, - [snd_soc_dapm_virt_mux] = 5, - [snd_soc_dapm_value_mux] = 5, - [snd_soc_dapm_dac] = 6, - [snd_soc_dapm_switch] = 7, - [snd_soc_dapm_mixer] = 7, - [snd_soc_dapm_mixer_named_ctl] = 7, - [snd_soc_dapm_pga] = 8, - [snd_soc_dapm_adc] = 9, - [snd_soc_dapm_out_drv] = 10, - [snd_soc_dapm_hp] = 10, - [snd_soc_dapm_spk] = 10, - [snd_soc_dapm_line] = 10, - [snd_soc_dapm_kcontrol] = 11, - [snd_soc_dapm_post] = 12, + [snd_soc_dapm_dai_in] = 4, + [snd_soc_dapm_dai_out] = 4, + [snd_soc_dapm_aif_in] = 4, + [snd_soc_dapm_aif_out] = 4, + [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_mux] = 6, + [snd_soc_dapm_virt_mux] = 6, + [snd_soc_dapm_value_mux] = 6, + [snd_soc_dapm_dac] = 7, + [snd_soc_dapm_switch] = 8, + [snd_soc_dapm_mixer] = 8, + [snd_soc_dapm_mixer_named_ctl] = 8, + [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_adc] = 10, + [snd_soc_dapm_out_drv] = 11, + [snd_soc_dapm_hp] = 11, + [snd_soc_dapm_spk] = 11, + [snd_soc_dapm_line] = 11, + [snd_soc_dapm_kcontrol] = 12, + [snd_soc_dapm_post] = 13, }; static int dapm_down_seq[] = { @@ -109,10 +109,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_dai_in] = 10, [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, - [snd_soc_dapm_clock_supply] = 12, - [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, - [snd_soc_dapm_post] = 13, + [snd_soc_dapm_clock_supply] = 13, + [snd_soc_dapm_regulator_supply] = 13, + [snd_soc_dapm_post] = 14, }; static void pop_wait(u32 pop_time) -- cgit From c6452e39e8286b88872aee20a4d083cfa65516bc Mon Sep 17 00:00:00 2001 From: Steffen Trumtrar Date: Fri, 11 Oct 2013 12:28:13 +0200 Subject: ASoC: mc13783: add mixer controls Add more controls to the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05b8e96..2b62737bf3d4 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -541,8 +541,26 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + + SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), + SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), + SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + + SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + + SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- cgit From bb7838d4f13c50df8ef7324f5fd4aeb729269e22 Mon Sep 17 00:00:00 2001 From: Steffen Trumtrar Date: Fri, 11 Oct 2013 12:28:14 +0200 Subject: ASoC: mc13783: add more DAPM routes Add more infrastructure (i.e. routes, muxes, switches) to the mc13783 DAPM. Signed-off-by: Steffen Trumtrar Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 52 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 2b62737bf3d4..f5472adee674 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -427,6 +427,29 @@ static const struct snd_kcontrol_new right_input_mux = static const struct snd_kcontrol_new samp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); +static const char * const speaker_amp_source_text[] = { + "CODEC", "Right" +}; +static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, + speaker_amp_source_text); +static const struct snd_kcontrol_new speaker_amp_source_mux = + SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); + +static const char * const headset_amp_source_text[] = { + "CODEC", "Mixer" +}; + +static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, + headset_amp_source_text); +static const struct snd_kcontrol_new headset_amp_source_mux = + SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); + +static const struct snd_kcontrol_new cdcout_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0); + +static const struct snd_kcontrol_new adc_bypass_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_CODEC, 16, 1, 0); + static const struct snd_kcontrol_new lamp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); @@ -464,12 +487,22 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, &right_input_mux), + SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0, + &speaker_amp_source_mux), + + SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0, + &headset_amp_source_mux), + SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), + SND_SOC_DAPM_PGA("Voice CODEC PGA", MC13783_AUDIO_RX1, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Voice CODEC Bypass", MC13783_AUDIO_CODEC, 16, 0, + &adc_bypass_ctl), + /* Output */ SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), @@ -477,10 +510,15 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), SND_SOC_DAPM_OUTPUT("HSR"), + SND_SOC_DAPM_OUTPUT("LSPL"), SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), + SND_SOC_DAPM_OUTPUT("CDCOUT"), - SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, + &cdcout_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, + &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, &hlamp_ctl), @@ -515,20 +553,28 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "ADC", NULL, "PGA Right Input"}, { "ADC", NULL, "ADC_Reset"}, + { "Voice CODEC PGA", "Voice CODEC Bypass", "ADC" }, + + { "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Speaker Amp Source MUX", "Right", "DAC PGA"}, + + { "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Headset Amp Source MUX", "Mixer", "DAC PGA"}, + /* Output */ { "HSL", NULL, "Headset Amp Left" }, { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, - { "SP", NULL, "Speaker Amp"}, - { "Speaker Amp", NULL, "DAC PGA"}, - { "LSP", NULL, "DAC PGA"}, - { "Headset Amp Left", NULL, "DAC PGA"}, - { "Headset Amp Right", NULL, "DAC PGA"}, + { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, + { "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"}, + { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, + { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, { "Line out Amp Left", NULL, "DAC PGA"}, { "Line out Amp Right", NULL, "DAC PGA"}, { "DAC PGA", NULL, "DAC"}, { "DAC", NULL, "DAC_E"}, + { "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"}, }; static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", -- cgit From 740ad6c328823f066efb8b907576a54ef92aca69 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Oct 2013 00:06:34 -0700 Subject: ASoC: rcar: fixup rsnd_platform_call() return value Un-implemented platform callback is not error. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fc83f0f2aead..28c24fcf8bc7 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -103,7 +103,7 @@ * rsnd_platform functions */ #define rsnd_platform_call(priv, dai, func, param...) \ - (!(priv->info->func) ? -ENODEV : \ + (!(priv->info->func) ? 0 : \ priv->info->func(param)) /* -- cgit From 2192f81c53a7879c803f0f7d6c49645fdf6c2f6a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Oct 2013 00:07:48 -0700 Subject: ASoC: rcar: add ID check on rsnd_dai_get() checking id in rsnd_dai_get() is good idea Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 28c24fcf8bc7..b234ed663073 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -318,6 +318,9 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) { + if ((id < 0) || (id >= rsnd_dai_nr(priv))) + return NULL; + return priv->rdai + id; } -- cgit From 1d198f26c98e6501659d741d530f7b65e4b7aec3 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Tue, 8 Oct 2013 15:55:45 -0700 Subject: sound: Remove unnecessary semicolons These aren't necessary after switch and if blocks. Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/oss/sb_ess.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/twl6040.c | 4 ++-- sound/soc/fsl/fsl_spdif.c | 4 ++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 6 +++--- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- 12 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index c0be085e4a20..0e7254bde4c2 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -1544,7 +1544,7 @@ static int ess_has_rec_mixer (int submodel) return 1; default: return 0; - }; + } }; #ifdef FKS_LOGGING diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index b46dc9b24dbd..9fb03b4ea925 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -671,7 +671,7 @@ static int snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return err; break; #endif - }; + } if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_SPDIF) { for (i = 0; i < ARRAY_SIZE(snd_vortex_mixer_spdif); i++) { diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index bb53dea85b17..8697cedccd21 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -777,7 +777,7 @@ static int asoc_ssc_init(struct device *dev) if (ret) { dev_err(dev, "Could not register PCM: %d\n", ret); goto err_unregister_dai; - }; + } return 0; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 5f9af1fb76e8..49cc5f6d6dba 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -328,7 +328,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ak4641->playback_fs = rate; ak4641_set_deemph(codec); - }; + } return 0; } diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index ea141e1d6f28..154433084708 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -382,7 +382,7 @@ static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai, break; default: return -EINVAL; - }; + } snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88f7204..e29cdb7ee232 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -429,7 +429,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, default: dev_err(codec->dev, "Invalid bit width\n"); return -EINVAL; - }; + } ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); if (ret < 0) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb6c323..35059a242fa4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e849ce..c0fea02114e1 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 52af7f6fb37f..364bf6a907e1 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -297,7 +297,7 @@ static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) @@ -310,7 +310,7 @@ static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) @@ -321,7 +321,7 @@ static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra20_i2s_regmap_config = { diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 551b3c93ce93..08bc6931c7c7 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -213,7 +213,7 @@ static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) @@ -234,7 +234,7 @@ static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) @@ -247,7 +247,7 @@ static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra20_spdif_regmap_config = { diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d554d46d08b5..805b1f399dc3 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -346,7 +346,7 @@ static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: break; - }; + } if (REG_IN_ARRAY(reg, CHANNEL_CTRL) || REG_IN_ARRAY(reg, CHANNEL_CLEAR) || @@ -381,7 +381,7 @@ static bool tegra30_ahub_apbif_volatile_reg(struct device *dev, return true; default: break; - }; + } if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) || REG_IN_ARRAY(reg, CHANNEL_STATUS) || diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 47565fd04505..bd71145f3639 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -369,7 +369,7 @@ static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) @@ -382,7 +382,7 @@ static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra30_i2s_regmap_config = { -- cgit From 1b3ed70a1b22e54b6adaf6ffebe1aa6f26465bae Mon Sep 17 00:00:00 2001 From: Felipe Pena Date: Sat, 12 Oct 2013 19:35:06 -0300 Subject: ASoC: fsl: Fix memory leak in imx-audmux.c When audmux_clk is used and clk_prepare_enable function succeed, the memory alloc'd to buf variable is leaked Signed-off-by: Felipe Pena Reviewed-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d3bf71a0ec56..ac869931d7f1 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -66,13 +66,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; int port = (int)file->private_data; u32 pdcr, ptcr; - if (!buf) - return -ENOMEM; - if (audmux_clk) { ret = clk_prepare_enable(audmux_clk); if (ret) @@ -85,6 +82,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (audmux_clk) clk_disable_unprepare(audmux_clk); + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); -- cgit From 48afa793525800eff66a2e792037108b7f0d8613 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Sun, 13 Oct 2013 18:17:50 +0200 Subject: ASoC: atmel: don't use devm_pinctrl_get_select_default() in probe Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for setting the default pins. Signed-off-by: Wolfram Sang Acked-by: Linus Walleij (personally at LCE13) Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_wm8904.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 7222380131ea..b4e36901a40b 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -12,7 +12,6 @@ #include #include #include -#include #include @@ -155,15 +154,8 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev) struct snd_soc_card *card = &atmel_asoc_wm8904_card; struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; struct clk *clk_src; - struct pinctrl *pinctrl; int id, ret; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "failed to request pinctrl\n"); - return PTR_ERR(pinctrl); - } - card->dev = &pdev->dev; ret = atmel_asoc_wm8904_dt_init(pdev); if (ret) { -- cgit From 67093b253bb09276420c7b0a311425705d2714b4 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Sun, 13 Oct 2013 21:51:26 +0200 Subject: ASoC: fsl: remove leftover release_mem_region When converting this driver to devm_ioremap_resource, the removal of this now unneeded function has been forgotten. Signed-off-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f58bcd85c07f..02722897914d 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -615,7 +615,6 @@ failed_pcm_dma: failed_pcm_fiq: snd_soc_unregister_component(&pdev->dev); failed_register: - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: snd_soc_set_ac97_ops(NULL); @@ -625,7 +624,6 @@ failed_clk: static int imx_ssi_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); imx_pcm_dma_exit(pdev); @@ -636,7 +634,6 @@ static int imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); snd_soc_set_ac97_ops(NULL); -- cgit From 5e049fce368dfe07702c3664add9ae7b45df1a9a Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 11 Oct 2013 15:43:17 -0600 Subject: ASoC: tegra: support new register layouts in Tegra124 Tegra124 introduces some small changes to the layout of some registers. Modify the affected drivers to program those registers appropriately based on which SoC they're running on. Tegra124 also introduced some new modules on the AHUB configlink register bus. These will require new entries in configlink_clocks[] in the AHUB driver. However, supporting that change likely relies on switching Tegra to the common reset framework, so I'll defer that change for now. Based-on-work-by: Arun Shamanna Lakshmi Based-on-work-by: Songhee Baek Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_ahub.c | 115 ++++++++++++++++++++++++++++++++----- sound/soc/tegra/tegra30_ahub.h | 38 +++++++++++- sound/soc/tegra/tegra30_i2s.c | 51 +++++++++++----- sound/soc/tegra/tegra30_i2s.h | 7 +++ sound/soc/tegra/tegra_asoc_utils.c | 2 + sound/soc/tegra/tegra_asoc_utils.h | 1 + 6 files changed, 186 insertions(+), 28 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d554d46d08b5..bdd19db4a08b 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -100,6 +100,7 @@ int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, { int channel; u32 reg, val; + struct tegra30_ahub_cif_conf cif_conf; channel = find_first_zero_bit(ahub->rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); @@ -123,15 +124,21 @@ int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16; tegra30_apbif_write(reg, val); + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; + reg = TEGRA30_AHUB_CIF_RX_CTRL + (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; - tegra30_apbif_write(reg, val); + ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf); return 0; } @@ -183,6 +190,7 @@ int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, { int channel; u32 reg, val; + struct tegra30_ahub_cif_conf cif_conf; channel = find_first_zero_bit(ahub->tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); @@ -206,15 +214,21 @@ int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16; tegra30_apbif_write(reg, val); + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; + reg = TEGRA30_AHUB_CIF_TX_CTRL + (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - tegra30_apbif_write(reg, val); + ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf); return 0; } @@ -437,13 +451,21 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { static struct tegra30_ahub_soc_data soc_data_tegra30 = { .clk_list_mask = CLK_LIST_MASK_TEGRA30, + .set_audio_cif = tegra30_ahub_set_cif, }; static struct tegra30_ahub_soc_data soc_data_tegra114 = { .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .set_audio_cif = tegra30_ahub_set_cif, +}; + +static struct tegra30_ahub_soc_data soc_data_tegra124 = { + .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .set_audio_cif = tegra124_ahub_set_cif, }; static const struct of_device_id tegra30_ahub_of_match[] = { + { .compatible = "nvidia,tegra124-ahub", .data = &soc_data_tegra124 }, { .compatible = "nvidia,tegra114-ahub", .data = &soc_data_tegra114 }, { .compatible = "nvidia,tegra30-ahub", .data = &soc_data_tegra30 }, {}, @@ -497,6 +519,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ahub); + ahub->soc_data = soc_data; ahub->dev = &pdev->dev; ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); @@ -669,6 +692,70 @@ static struct platform_driver tegra30_ahub_driver = { }; module_platform_driver(tegra30_ahub_driver); +void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf) +{ + unsigned int value; + + value = (conf->threshold << + TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + ((conf->audio_channels - 1) << + TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + ((conf->client_channels - 1) << + TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + (conf->audio_bits << + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) | + (conf->client_bits << + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) | + (conf->expand << + TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) | + (conf->stereo_conv << + TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) | + (conf->replicate << + TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) | + (conf->direction << + TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) | + (conf->truncate << + TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) | + (conf->mono_conv << + TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT); + + regmap_write(regmap, reg, value); +} +EXPORT_SYMBOL_GPL(tegra30_ahub_set_cif); + +void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf) +{ + unsigned int value; + + value = (conf->threshold << + TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + ((conf->audio_channels - 1) << + TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + ((conf->client_channels - 1) << + TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + (conf->audio_bits << + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) | + (conf->client_bits << + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) | + (conf->expand << + TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) | + (conf->stereo_conv << + TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) | + (conf->replicate << + TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) | + (conf->direction << + TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) | + (conf->truncate << + TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) | + (conf->mono_conv << + TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT); + + regmap_write(regmap, reg, value); +} +EXPORT_SYMBOL_GPL(tegra124_ahub_set_cif); + MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra30 AHUB driver"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h index 09766cdc45ca..d67321d90faa 100644 --- a/sound/soc/tegra/tegra30_ahub.h +++ b/sound/soc/tegra/tegra30_ahub.h @@ -25,16 +25,30 @@ #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0xf #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT 24 +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0x3f +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) + /* Channel count minus 1 */ #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 24 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 7 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) +/* Channel count minus 1 */ +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 20 +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 0xf +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) + /* Channel count minus 1 */ #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 7 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) +/* Channel count minus 1 */ +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 0xf +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) + #define TEGRA30_AUDIOCIF_BITS_4 0 #define TEGRA30_AUDIOCIF_BITS_8 1 #define TEGRA30_AUDIOCIF_BITS_12 2 @@ -86,7 +100,7 @@ #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1 (TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG (TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) -#define TEGRA30_AUDIOCIF_CTRL_REPLICATE 3 +#define TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT 3 #define TEGRA30_AUDIOCIF_DIRECTION_TX 0 #define TEGRA30_AUDIOCIF_DIRECTION_RX 1 @@ -468,8 +482,30 @@ extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, enum tegra30_ahub_txcif txcif); extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); +struct tegra30_ahub_cif_conf { + unsigned int threshold; + unsigned int audio_channels; + unsigned int client_channels; + unsigned int audio_bits; + unsigned int client_bits; + unsigned int expand; + unsigned int stereo_conv; + unsigned int replicate; + unsigned int direction; + unsigned int truncate; + unsigned int mono_conv; +}; + +void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf); +void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf); + struct tegra30_ahub_soc_data { u32 clk_list_mask; + void (*set_audio_cif)(struct regmap *regmap, + unsigned int reg, + struct tegra30_ahub_cif_conf *conf); /* * FIXME: There are many more differences in HW, such as: * - More APBIF channels. diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 47565fd04505..5f20b695eba2 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include @@ -179,6 +180,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); unsigned int mask, val, reg; int ret, sample_size, srate, i2sclock, bitcnt; + struct tegra30_ahub_cif_conf cif_conf; if (params_channels(params) != 2) return -EINVAL; @@ -217,21 +219,26 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16; + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX; reg = TEGRA30_I2S_CIF_RX_CTRL; } else { - val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX; reg = TEGRA30_I2S_CIF_TX_CTRL; } - regmap_write(i2s->regmap, reg, val); + i2s->soc_data->set_audio_cif(i2s->regmap, reg, &cif_conf); val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); @@ -396,9 +403,24 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .cache_type = REGCACHE_RBTREE, }; +static const struct tegra30_i2s_soc_data tegra30_i2s_config = { + .set_audio_cif = tegra30_ahub_set_cif, +}; + +static const struct tegra30_i2s_soc_data tegra124_i2s_config = { + .set_audio_cif = tegra124_ahub_set_cif, +}; + +static const struct of_device_id tegra30_i2s_of_match[] = { + { .compatible = "nvidia,tegra124-i2s", .data = &tegra124_i2s_config }, + { .compatible = "nvidia,tegra30-i2s", .data = &tegra30_i2s_config }, + {}, +}; + static int tegra30_i2s_platform_probe(struct platform_device *pdev) { struct tegra30_i2s *i2s; + const struct of_device_id *match; u32 cif_ids[2]; struct resource *mem, *memregion; void __iomem *regs; @@ -412,6 +434,14 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, i2s); + match = of_match_device(tegra30_i2s_of_match, &pdev->dev); + if (!match) { + dev_err(&pdev->dev, "Error: No device match found\n"); + ret = -ENODEV; + goto err; + } + i2s->soc_data = (struct tegra30_i2s_soc_data *)match->data; + i2s->dai = tegra30_i2s_dai_template; i2s->dai.name = dev_name(&pdev->dev); @@ -539,11 +569,6 @@ static int tegra30_i2s_resume(struct device *dev) } #endif -static const struct of_device_id tegra30_i2s_of_match[] = { - { .compatible = "nvidia,tegra30-i2s", }, - {}, -}; - static const struct dev_pm_ops tegra30_i2s_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, tegra30_i2s_runtime_resume, NULL) diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index bea23afe3b9f..4d0b0a30dbfb 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -225,7 +225,14 @@ #define TEGRA30_I2S_LCOEF_COEF_MASK_US 0xffff #define TEGRA30_I2S_LCOEF_COEF_MASK (TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT) +struct tegra30_i2s_soc_data { + void (*set_audio_cif)(struct regmap *regmap, + unsigned int reg, + struct tegra30_ahub_cif_conf *conf); +}; + struct tegra30_i2s { + const struct tegra30_i2s_soc_data *soc_data; struct snd_soc_dai_driver dai; int cif_id; struct clk *clk_i2s; diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index d173880f290d..1be311c51a18 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -182,6 +182,8 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; else if (of_machine_is_compatible("nvidia,tegra114")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; + else if (of_machine_is_compatible("nvidia,tegra124")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA124; else { dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 19fdcafed32f..9577121ce971 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -30,6 +30,7 @@ enum tegra_asoc_utils_soc { TEGRA_ASOC_UTILS_SOC_TEGRA20, TEGRA_ASOC_UTILS_SOC_TEGRA30, TEGRA_ASOC_UTILS_SOC_TEGRA114, + TEGRA_ASOC_UTILS_SOC_TEGRA124, }; struct tegra_asoc_utils_data { -- cgit From 9f1614aae59033b79941220f65ad36f5fe54a579 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 11 Oct 2013 12:11:02 +0200 Subject: ASoC: snd_soc_dai_ops trigger function description Add a comment to the trigger function in snd_soc_dai_ops struct about possible command sequences. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227d35d3..0f2e5daa1451 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -166,6 +166,13 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*prepare)(struct snd_pcm_substream *, struct snd_soc_dai *); + /* + * NOTE: Commands passed to the trigger function are not necessarily + * compatible with the current state of the dai. For example this + * sequence of commands is possible: START STOP STOP. + * So do not unconditionally use refcounting functions in the trigger + * function, e.g. clk_enable/disable. + */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); int (*bespoke_trigger)(struct snd_pcm_substream *, int, -- cgit From 88cf632a135188db35b4db412a06b155fa444eb1 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 11 Oct 2013 12:11:03 +0200 Subject: ASoC: mxs-saif: Store saif state Trigger commands may be passed multiple times. To avoid errors with clk_enable/disable, store the saif state and return if saif is already running/stopped. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 8 ++++++++ sound/soc/mxs/mxs-saif.h | 5 +++++ 2 files changed, 13 insertions(+) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b56b8a0e8deb..c8ead011c95b 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -503,6 +503,9 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (saif->state == MXS_SAIF_STATE_RUNNING) + return 0; + dev_dbg(cpu_dai->dev, "start\n"); clk_enable(master_saif->clk); @@ -543,6 +546,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } master_saif->ongoing = 1; + saif->state = MXS_SAIF_STATE_RUNNING; dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), @@ -555,6 +559,9 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (saif->state == MXS_SAIF_STATE_STOPPED) + return 0; + dev_dbg(cpu_dai->dev, "stop\n"); /* wait a while for the current sample to complete */ @@ -575,6 +582,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } master_saif->ongoing = 0; + saif->state = MXS_SAIF_STATE_STOPPED; break; default: diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 53eaa4bf0e27..fbaf7badfdfb 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -124,6 +124,11 @@ struct mxs_saif { u32 fifo_underrun; u32 fifo_overrun; + + enum { + MXS_SAIF_STATE_STOPPED, + MXS_SAIF_STATE_RUNNING, + } state; }; extern int mxs_saif_put_mclk(unsigned int saif_id); -- cgit From 863ebddec85c5ce2fb2e7742e8834a3bd69a2512 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 11 Oct 2013 12:11:04 +0200 Subject: ASoC: mxs-saif: Handle errors in trigger function Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c8ead011c95b..fc3d89b75d48 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -494,6 +494,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); struct mxs_saif *master_saif; u32 delay; + int ret; master_saif = mxs_saif_get_master(saif); if (!master_saif) @@ -508,21 +509,32 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(master_saif->clk); - if (!master_saif->mclk_in_use) - __raw_writel(BM_SAIF_CTRL_RUN, - master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + ret = clk_enable(master_saif->clk); + if (ret) { + dev_err(saif->dev, "Failed to enable master clock\n"); + return ret; + } /* * If the saif's master is not himself, we also need to enable * itself clk for its internal basic logic to work. */ if (saif != master_saif) { - clk_enable(saif->clk); + ret = clk_enable(saif->clk); + if (ret) { + dev_err(saif->dev, "Failed to enable master clock\n"); + clk_disable(master_saif->clk); + return ret; + } + __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); } + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* * write data to saif data register to trigger -- cgit From de0022d4bf3a5acaef75cf1c7c2f8d71b020e8c9 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Wed, 16 Oct 2013 15:58:48 +0530 Subject: ASoC: smdk_wm8994: Add .pm to struct smdk_audio_driver Register PM ops for this driver. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 831972d24fb9..b072bd107b31 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -9,6 +9,7 @@ #include "../codecs/wm8994.h" #include +#include #include #include #include @@ -206,6 +207,7 @@ static struct platform_driver smdk_audio_driver = { .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), + .pm = &snd_soc_pm_ops, }, .probe = smdk_audio_probe, }; -- cgit From a7ea1b7249adc8c090a0b277ab5f3737ee4023c1 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:56 +0530 Subject: ASoC: cs4271: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a20f1bb8f071..f6e953454bc0 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include -- cgit From 193a47162c93afa09fffd04a04443f14d402c606 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:57 +0530 Subject: ASoC: pcm1681: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 651ce0923675..54ea15b87bfc 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include -- cgit From 4b2fa5121c758db6fe9ed4931b54e390661395de Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:58 +0530 Subject: ASoC: pcm1792a: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 2a8eccf64c76..6f14c50a7f0f 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include "pcm1792a.h" -- cgit From 285d00c11b0a8d0ef63c176f88caab5071c9e80d Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:59 +0530 Subject: ASoC: tas5086: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 2996d2ea026b..fe4d29d88564 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -37,6 +37,7 @@ #include #include #include +#include #include #include #include -- cgit From b3b70786ec18ef3088b55b76258bbd48d75aee08 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:24:00 +0530 Subject: ASoC: tlv320aic3x: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 892c108ca67a..f8b9fa6b6f0a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include -- cgit From 8aa99652cd52af07f4fa49fc50d78ede48c9c9b3 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:24:01 +0530 Subject: ASoC: atmel: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 802717eccbd0..f15bff1548f8 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -37,6 +37,7 @@ #include #include #include +#include #include -- cgit From 420f9739a62cdb027f5580d25c813501ff93aa6f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 16 Oct 2013 23:10:31 +0200 Subject: thinkpad-acpi: Add mute and mic-mute LED functionality The LEDs are currently not visible to userspace, for security reasons. They are exported through thinkpad_acpi.h for use by the snd-hda-intel driver. Thanks to Alex Hung and Takashi Iwai for writing parts of this patch. Signed-off-by: David Henningsson Acked-by: Henrique de Moraes Holschuh Signed-off-by: Takashi Iwai --- Documentation/laptops/thinkpad-acpi.txt | 7 ++- drivers/platform/x86/thinkpad_acpi.c | 92 ++++++++++++++++++++++++++++++++- include/linux/thinkpad_acpi.h | 15 ++++++ 3 files changed, 111 insertions(+), 3 deletions(-) create mode 100644 include/linux/thinkpad_acpi.h diff --git a/Documentation/laptops/thinkpad-acpi.txt b/Documentation/laptops/thinkpad-acpi.txt index 86c52360ffe7..fc04c14de4bb 100644 --- a/Documentation/laptops/thinkpad-acpi.txt +++ b/Documentation/laptops/thinkpad-acpi.txt @@ -1,7 +1,7 @@ ThinkPad ACPI Extras Driver - Version 0.24 - December 11th, 2009 + Version 0.25 + October 16th, 2013 Borislav Deianov Henrique de Moraes Holschuh @@ -741,6 +741,9 @@ compiled with the CONFIG_THINKPAD_ACPI_UNSAFE_LEDS option enabled. Distributions must never enable this option. Individual users that are aware of the consequences are welcome to enabling it. +Audio mute and microphone mute LEDs are supported, but currently not +visible to userspace. They are used by the snd-hda-intel audio driver. + procfs notes: The available commands are: diff --git a/drivers/platform/x86/thinkpad_acpi.c b/drivers/platform/x86/thinkpad_acpi.c index 03ca6c139f1a..0b7efb269cf1 100644 --- a/drivers/platform/x86/thinkpad_acpi.c +++ b/drivers/platform/x86/thinkpad_acpi.c @@ -23,7 +23,7 @@ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt -#define TPACPI_VERSION "0.24" +#define TPACPI_VERSION "0.25" #define TPACPI_SYSFS_VERSION 0x020700 /* @@ -88,6 +88,7 @@ #include +#include /* ThinkPad CMOS commands */ #define TP_CMOS_VOLUME_DOWN 0 @@ -8350,6 +8351,91 @@ static struct ibm_struct fan_driver_data = { .resume = fan_resume, }; +/************************************************************************* + * Mute LED subdriver + */ + + +struct tp_led_table { + acpi_string name; + int on_value; + int off_value; + int state; +}; + +static struct tp_led_table led_tables[] = { + [TPACPI_LED_MUTE] = { + .name = "SSMS", + .on_value = 1, + .off_value = 0, + }, + [TPACPI_LED_MICMUTE] = { + .name = "MMTS", + .on_value = 2, + .off_value = 0, + }, +}; + +static int mute_led_on_off(struct tp_led_table *t, bool state) +{ + acpi_handle temp; + int output; + + if (!ACPI_SUCCESS(acpi_get_handle(hkey_handle, t->name, &temp))) { + pr_warn("Thinkpad ACPI has no %s interface.\n", t->name); + return -EIO; + } + + if (!acpi_evalf(hkey_handle, &output, t->name, "dd", + state ? t->on_value : t->off_value)) + return -EIO; + + t->state = state; + return state; +} + +int tpacpi_led_set(int whichled, bool on) +{ + struct tp_led_table *t; + + if (whichled < 0 || whichled >= TPACPI_LED_MAX) + return -EINVAL; + + t = &led_tables[whichled]; + if (t->state < 0 || t->state == on) + return t->state; + return mute_led_on_off(t, on); +} +EXPORT_SYMBOL_GPL(tpacpi_led_set); + +static int mute_led_init(struct ibm_init_struct *iibm) +{ + acpi_handle temp; + int i; + + for (i = 0; i < TPACPI_LED_MAX; i++) { + struct tp_led_table *t = &led_tables[i]; + if (ACPI_SUCCESS(acpi_get_handle(hkey_handle, t->name, &temp))) + mute_led_on_off(t, false); + else + t->state = -ENODEV; + } + return 0; +} + +static void mute_led_exit(void) +{ + int i; + + for (i = 0; i < TPACPI_LED_MAX; i++) + tpacpi_led_set(i, false); +} + +static struct ibm_struct mute_led_driver_data = { + .name = "mute_led", + .exit = mute_led_exit, +}; + /**************************************************************************** **************************************************************************** * @@ -8768,6 +8854,10 @@ static struct ibm_init_struct ibms_init[] __initdata = { .init = fan_init, .data = &fan_driver_data, }, + { + .init = mute_led_init, + .data = &mute_led_driver_data, + }, }; static int __init set_ibm_param(const char *val, struct kernel_param *kp) diff --git a/include/linux/thinkpad_acpi.h b/include/linux/thinkpad_acpi.h new file mode 100644 index 000000000000..361de59a2285 --- /dev/null +++ b/include/linux/thinkpad_acpi.h @@ -0,0 +1,15 @@ +#ifndef __THINKPAD_ACPI_H__ +#define __THINKPAD_ACPI_H__ + +/* These two functions return 0 if success, or negative error code + (e g -ENODEV if no led present) */ + +enum { + TPACPI_LED_MUTE, + TPACPI_LED_MICMUTE, + TPACPI_LED_MAX, +}; + +int tpacpi_led_set(int whichled, bool on); + +#endif -- cgit From 360fec281e2012b405b5fac48449f69e4a049949 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 16 Oct 2013 23:10:32 +0200 Subject: ALSA: hda - add HDA_FIXUP_ACT_FREE action A fixup which should be called before codec being freed will come to use in the next patch. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 2e7493ef8ee0..a71bf348e389 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -428,6 +428,7 @@ enum { HDA_FIXUP_ACT_PROBE, HDA_FIXUP_ACT_INIT, HDA_FIXUP_ACT_BUILD, + HDA_FIXUP_ACT_FREE, }; int snd_hda_add_verbs(struct hda_codec *codec, const struct hda_verb *list); -- cgit From 08cf680ccafd5df9885fbcd0cab85221df00b44b Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 16 Oct 2013 23:10:33 +0200 Subject: ALSA: hda - add connection to thinkpad_acpi to control mute/micmute LEDs Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 92 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 90 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ec68eaea0336..993b25c17711 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3208,11 +3208,17 @@ static int cx_auto_init(struct hda_codec *codec) return 0; } +static void cx_auto_free(struct hda_codec *codec) +{ + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_FREE); + snd_hda_gen_free(codec); +} + static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = cx_auto_init, - .free = snd_hda_gen_free, + .free = cx_auto_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, @@ -3232,8 +3238,84 @@ enum { CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, CXT_FIXUP_GPIO1, + CXT_FIXUP_THINKPAD_ACPI, }; +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + +#include + +static int (*led_set_func)(int, bool); + +static void update_tpacpi_mute_led(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct conexant_spec *spec = codec->spec; + + if (spec->dynamic_eapd) + cx_auto_vmaster_hook(private_data, enabled); + + if (led_set_func) + led_set_func(TPACPI_LED_MUTE, !enabled); +} + +static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + if (!ucontrol || !led_set_func) + return; + if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { + /* TODO: How do I verify if it's a mono or stereo here? */ + bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; + led_set_func(TPACPI_LED_MICMUTE, !val); + } +} + +static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!led_set_func) + led_set_func = symbol_request(tpacpi_led_set); + if (!led_set_func) { + snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); + return; + } + + removefunc = true; + if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + spec->gen.vmaster_mute.hook = update_tpacpi_mute_led; + removefunc = false; + } + if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { + if (spec->gen.num_adc_nids > 1) + snd_printdd("Skipping micmute LED control due to several ADCs"); + else { + spec->gen.cap_sync_hook = update_tpacpi_micmute_led; + removefunc = false; + } + } + } + + if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(tpacpi_led_set); + led_set_func = NULL; + } +} + +#else + +static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif + static void cxt_fixup_stereo_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3344,6 +3426,8 @@ static const struct hda_fixup cxt_fixups[] = { [CXT_PINCFG_LENOVO_TP410] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lenovo_tp410, + .chained = true, + .chain_id = CXT_FIXUP_THINKPAD_ACPI, }, [CXT_PINCFG_LEMOTE_A1004] = { .type = HDA_FIXUP_PINS, @@ -3385,6 +3469,10 @@ static const struct hda_fixup cxt_fixups[] = { { } }, }, + [CXT_FIXUP_THINKPAD_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_thinkpad_acpi, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3507,7 +3595,7 @@ static int patch_conexant_auto(struct hda_codec *codec) return 0; error: - snd_hda_gen_free(codec); + cx_auto_free(codec); return err; } -- cgit From 4758fed912d7cd0ba53d2694e89b884114de6580 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Oct 2013 17:56:25 +0200 Subject: ALSA: hda - Treat zero connection as non-error The zero-length connection list happens so often on Haswell HDMI, and it results in warning messages like ALSA: hda_codec: invalid CONNECT_LIST verb 5[1]:0 at each time the codec resumes from the power-save, which is fairly annoying. Since this is no real error, make it shown only in the verbose debug mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 68801ba001ce..a1632f4056d4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -565,7 +565,7 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, range_val = !!(parm & (1 << (shift-1))); /* ranges */ val = parm & mask; if (val == 0 && null_count++) { /* no second chance */ - snd_printk(KERN_WARNING "hda_codec: " + snd_printdd("hda_codec: " "invalid CONNECT_LIST verb %x[%i]:%x\n", nid, i, parm); return 0; -- cgit From cbbaa603a03cc46681e24d6b2804b62fde95a2af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Oct 2013 18:03:24 +0200 Subject: ALSA: hda - Fix possible races in HDMI driver Some per_pin fields and ELD contents might be changed dynamically in multiple ways where the concurrent accesses are still opened in the current code. This patch fixes such possible races by using eld->lock in appropriate places. Reported-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b899eba6f279..c2bad9587d54 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1342,6 +1342,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) bool update_eld = false; bool eld_changed = false; + mutex_lock(&pin_eld->lock); pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); if (pin_eld->monitor_present) eld->eld_valid = !!(present & AC_PINSENSE_ELDV); @@ -1371,11 +1372,10 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) queue_delayed_work(codec->bus->workq, &per_pin->work, msecs_to_jiffies(300)); - return; + goto unlock; } } - mutex_lock(&pin_eld->lock); if (pin_eld->eld_valid && !eld->eld_valid) { update_eld = true; eld_changed = true; @@ -1400,12 +1400,13 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } - mutex_unlock(&pin_eld->lock); if (eld_changed) snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &per_pin->eld_ctl->id); + unlock: + mutex_unlock(&pin_eld->lock); } static void hdmi_repoll_eld(struct work_struct *work) @@ -1576,10 +1577,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); + mutex_lock(&per_pin->sink_eld.lock); per_pin->channels = substream->runtime->channels; per_pin->setup = true; hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); + mutex_unlock(&per_pin->sink_eld.lock); return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1617,11 +1620,14 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_pin = get_pin(spec, pin_idx); snd_hda_spdif_ctls_unassign(codec, pin_idx); + + mutex_lock(&per_pin->sink_eld.lock); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); per_pin->setup = false; per_pin->channels = 0; + mutex_unlock(&per_pin->sink_eld.lock); } return 0; @@ -1750,10 +1756,12 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); if (ca < 0) return -EINVAL; + mutex_lock(&per_pin->sink_eld.lock); per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + mutex_unlock(&per_pin->sink_eld.lock); return 0; } -- cgit From a4e9a38b40a0e2f7dad1a0b355896d23fbdd16e0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Oct 2013 18:21:12 +0200 Subject: ALSA: hda - Move mutex from hda_eld to per_pin in HDMI codec driver Since the lock is used primarily in patch_hdmi.c, it's better to move it in the local struct instead of exporting in hda_eld. The only functions requiring the lock in hda_eld.c are proc accessors. So in this patch, the proc entry and its creation/deletion/accessors are moved into patch_hdmi.c, together with the mutex lock to pin_spec struct. The former proc info functions are exported so that they can be called from patch_hdmi.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 48 +++----------------- sound/pci/hda/hda_local.h | 22 ++------- sound/pci/hda/patch_hdmi.c | 108 ++++++++++++++++++++++++++++++++++++--------- 3 files changed, 97 insertions(+), 81 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index d0d7ac1e99d2..f62356c2f54c 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -478,10 +478,9 @@ static void hdmi_print_sad_info(int i, struct cea_sad *a, snd_iprintf(buffer, "sad%d_profile\t\t%d\n", i, a->profile); } -static void hdmi_print_eld_info(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) +void snd_hdmi_print_eld_info(struct hdmi_eld *eld, + struct snd_info_buffer *buffer) { - struct hdmi_eld *eld = entry->private_data; struct parsed_hdmi_eld *e = &eld->info; char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; int i; @@ -500,13 +499,10 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry, [4 ... 7] = "reserved" }; - mutex_lock(&eld->lock); snd_iprintf(buffer, "monitor_present\t\t%d\n", eld->monitor_present); snd_iprintf(buffer, "eld_valid\t\t%d\n", eld->eld_valid); - if (!eld->eld_valid) { - mutex_unlock(&eld->lock); + if (!eld->eld_valid) return; - } snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name); snd_iprintf(buffer, "connection_type\t\t%s\n", eld_connection_type_names[e->conn_type]); @@ -528,13 +524,11 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry, for (i = 0; i < e->sad_count; i++) hdmi_print_sad_info(i, e->sad + i, buffer); - mutex_unlock(&eld->lock); } -static void hdmi_write_eld_info(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) +void snd_hdmi_write_eld_info(struct hdmi_eld *eld, + struct snd_info_buffer *buffer) { - struct hdmi_eld *eld = entry->private_data; struct parsed_hdmi_eld *e = &eld->info; char line[64]; char name[64]; @@ -542,7 +536,6 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, long long val; unsigned int n; - mutex_lock(&eld->lock); while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%s %llx", name, &val) != 2) continue; @@ -594,38 +587,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, e->sad_count = n + 1; } } - mutex_unlock(&eld->lock); -} - - -int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, - int index) -{ - char name[32]; - struct snd_info_entry *entry; - int err; - - snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index); - err = snd_card_proc_new(codec->bus->card, name, &entry); - if (err < 0) - return err; - - snd_info_set_text_ops(entry, eld, hdmi_print_eld_info); - entry->c.text.write = hdmi_write_eld_info; - entry->mode |= S_IWUSR; - eld->proc_entry = entry; - - return 0; -} - -void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) -{ - if (!codec->bus->shutdown && eld->proc_entry) { - snd_device_free(codec->bus->card, eld->proc_entry); - eld->proc_entry = NULL; - } } - #endif /* CONFIG_PROC_FS */ /* update PCM info based on ELD */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index a71bf348e389..46cddd4c7b72 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -752,10 +752,6 @@ struct hdmi_eld { int eld_size; char eld_buffer[ELD_MAX_SIZE]; struct parsed_hdmi_eld info; - struct mutex lock; -#ifdef CONFIG_PROC_FS - struct snd_info_entry *proc_entry; -#endif }; int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid); @@ -768,20 +764,10 @@ void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, struct hda_pcm_stream *hinfo); #ifdef CONFIG_PROC_FS -int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, - int index); -void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld); -#else -static inline int snd_hda_eld_proc_new(struct hda_codec *codec, - struct hdmi_eld *eld, - int index) -{ - return 0; -} -static inline void snd_hda_eld_proc_free(struct hda_codec *codec, - struct hdmi_eld *eld) -{ -} +void snd_hdmi_print_eld_info(struct hdmi_eld *eld, + struct snd_info_buffer *buffer); +void snd_hdmi_write_eld_info(struct hdmi_eld *eld, + struct snd_info_buffer *buffer); #endif #define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c2bad9587d54..804adb872e63 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -67,6 +67,7 @@ struct hdmi_spec_per_pin { struct hda_codec *codec; struct hdmi_eld sink_eld; + struct mutex lock; struct delayed_work work; struct snd_kcontrol *eld_ctl; int repoll_count; @@ -76,6 +77,9 @@ struct hdmi_spec_per_pin { bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ char pcm_name[8]; /* filled in build_pcm callbacks */ +#ifdef CONFIG_PROC_FS + struct snd_info_entry *proc_entry; +#endif }; struct hdmi_spec { @@ -349,17 +353,19 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; int pin_idx; uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; pin_idx = kcontrol->private_value; - eld = &get_pin(spec, pin_idx)->sink_eld; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; - mutex_lock(&eld->lock); + mutex_lock(&per_pin->lock); uinfo->count = eld->eld_valid ? eld->eld_size : 0; - mutex_unlock(&eld->lock); + mutex_unlock(&per_pin->lock); return 0; } @@ -369,15 +375,17 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; int pin_idx; pin_idx = kcontrol->private_value; - eld = &get_pin(spec, pin_idx)->sink_eld; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; - mutex_lock(&eld->lock); + mutex_lock(&per_pin->lock); if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) { - mutex_unlock(&eld->lock); + mutex_unlock(&per_pin->lock); snd_BUG(); return -EINVAL; } @@ -387,7 +395,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, if (eld->eld_valid) memcpy(ucontrol->value.bytes.data, eld->eld_buffer, eld->eld_size); - mutex_unlock(&eld->lock); + mutex_unlock(&per_pin->lock); return 0; } @@ -478,6 +486,68 @@ static void hdmi_set_channel_count(struct hda_codec *codec, AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); } +/* + * ELD proc files + */ + +#ifdef CONFIG_PROC_FS +static void print_eld_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hdmi_spec_per_pin *per_pin = entry->private_data; + + mutex_lock(&per_pin->lock); + snd_hdmi_print_eld_info(&per_pin->sink_eld, buffer); + mutex_unlock(&per_pin->lock); +} + +static void write_eld_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hdmi_spec_per_pin *per_pin = entry->private_data; + + mutex_lock(&per_pin->lock); + snd_hdmi_write_eld_info(&per_pin->sink_eld, buffer); + mutex_unlock(&per_pin->lock); +} + +static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) +{ + char name[32]; + struct hda_codec *codec = per_pin->codec; + struct snd_info_entry *entry; + int err; + + snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index); + err = snd_card_proc_new(codec->bus->card, name, &entry); + if (err < 0) + return err; + + snd_info_set_text_ops(entry, per_pin, print_eld_info); + entry->c.text.write = write_eld_info; + entry->mode |= S_IWUSR; + per_pin->proc_entry = entry; + + return 0; +} + +static void eld_proc_free(struct hdmi_spec_per_pin *per_pin) +{ + if (!per_pin->codec->bus->shutdown && per_pin->proc_entry) { + snd_device_free(per_pin->codec->bus->card, per_pin->proc_entry); + per_pin->proc_entry = NULL; + } +} +#else +static inline int snd_hda_eld_proc_new(struct hdmi_spec_per_pin *per_pin, + int index) +{ + return 0; +} +static inline void snd_hda_eld_proc_free(struct hdmi_spec_per_pin *per_pin) +{ +} +#endif /* * Channel mapping routines @@ -1342,7 +1412,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) bool update_eld = false; bool eld_changed = false; - mutex_lock(&pin_eld->lock); + mutex_lock(&per_pin->lock); pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); if (pin_eld->monitor_present) eld->eld_valid = !!(present & AC_PINSENSE_ELDV); @@ -1406,7 +1476,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &per_pin->eld_ctl->id); unlock: - mutex_unlock(&pin_eld->lock); + mutex_unlock(&per_pin->lock); } static void hdmi_repoll_eld(struct work_struct *work) @@ -1577,12 +1647,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); - mutex_lock(&per_pin->sink_eld.lock); + mutex_lock(&per_pin->lock); per_pin->channels = substream->runtime->channels; per_pin->setup = true; hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); - mutex_unlock(&per_pin->sink_eld.lock); + mutex_unlock(&per_pin->lock); return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1621,13 +1691,13 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pin_idx); - mutex_lock(&per_pin->sink_eld.lock); + mutex_lock(&per_pin->lock); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); per_pin->setup = false; per_pin->channels = 0; - mutex_unlock(&per_pin->sink_eld.lock); + mutex_unlock(&per_pin->lock); } return 0; @@ -1756,12 +1826,12 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); if (ca < 0) return -EINVAL; - mutex_lock(&per_pin->sink_eld.lock); + mutex_lock(&per_pin->lock); per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); - mutex_unlock(&per_pin->sink_eld.lock); + mutex_unlock(&per_pin->lock); return 0; } @@ -1878,12 +1948,11 @@ static int generic_hdmi_init_per_pins(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); - struct hdmi_eld *eld = &per_pin->sink_eld; per_pin->codec = codec; - mutex_init(&eld->lock); + mutex_init(&per_pin->lock); INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld); - snd_hda_eld_proc_new(codec, eld, pin_idx); + eld_proc_new(per_pin, pin_idx); } return 0; } @@ -1924,10 +1993,9 @@ static void generic_hdmi_free(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); - struct hdmi_eld *eld = &per_pin->sink_eld; cancel_delayed_work(&per_pin->work); - snd_hda_eld_proc_free(codec, eld); + eld_proc_free(per_pin); } flush_workqueue(codec->bus->workq); -- cgit From 82fbb4f7b47683077e0716474d4f1ce65a2146cb Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:04:49 +0200 Subject: ALSA: add DICE driver As a start point for further development, this is an incomplete driver for DICE devices: - only playback (so no clock source except the bus clock) - only 44.1 kHz - no MIDI - recovery after bus reset is slow - hwdep device is created, but not actually implemented Contains compilation fixes by Stefan Richter. Signed-off-by: Clemens Ladisch --- Documentation/ioctl/ioctl-number.txt | 1 + include/uapi/sound/Kbuild | 1 + include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 51 ++ sound/firewire/Kconfig | 13 + sound/firewire/Makefile | 2 + sound/firewire/dice.c | 1008 ++++++++++++++++++++++++++++++++++ 7 files changed, 1078 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/firewire.h create mode 100644 sound/firewire/dice.c diff --git a/Documentation/ioctl/ioctl-number.txt b/Documentation/ioctl/ioctl-number.txt index 2a5f0e14efa3..7cbfa3c4fc3d 100644 --- a/Documentation/ioctl/ioctl-number.txt +++ b/Documentation/ioctl/ioctl-number.txt @@ -138,6 +138,7 @@ Code Seq#(hex) Include File Comments 'H' C0-DF net/bluetooth/cmtp/cmtp.h conflict! 'H' C0-DF net/bluetooth/bnep/bnep.h conflict! 'H' F1 linux/hid-roccat.h +'H' F8-FA sound/firewire.h 'I' all linux/isdn.h conflict! 'I' 00-0F drivers/isdn/divert/isdn_divert.h conflict! 'I' 40-4F linux/mISDNif.h conflict! diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild index 0f7d279ebde3..a7f27704f980 100644 --- a/include/uapi/sound/Kbuild +++ b/include/uapi/sound/Kbuild @@ -5,6 +5,7 @@ header-y += asound_fm.h header-y += compress_offload.h header-y += compress_params.h header-y += emu10k1.h +header-y += firewire.h header-y += hdsp.h header-y += hdspm.h header-y += sb16_csp.h diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 041203f20f6d..9fc6219d3848 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -93,9 +93,10 @@ enum { SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */ SNDRV_HWDEP_IFACE_HDA, /* HD-audio */ SNDRV_HWDEP_IFACE_USB_STREAM, /* direct access to usb stream */ + SNDRV_HWDEP_IFACE_FW_DICE, /* TC DICE FireWire device */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_USB_STREAM + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_DICE }; struct snd_hwdep_info { diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h new file mode 100644 index 000000000000..e86131ca49e5 --- /dev/null +++ b/include/uapi/sound/firewire.h @@ -0,0 +1,51 @@ +#ifndef UAPI_SOUND_FIREWIRE_H_INCLUDED +#define UAPI_SOUND_FIREWIRE_H_INCLUDED + +#include + +/* events can be read() from the hwdep device */ + +#define SNDRV_FIREWIRE_EVENT_LOCK_STATUS 0x000010cc +#define SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION 0xd1ce004e + +struct snd_firewire_event_common { + unsigned int type; /* SNDRV_FIREWIRE_EVENT_xxx */ +}; + +struct snd_firewire_event_lock_status { + unsigned int type; + unsigned int status; /* 0/1 = unlocked/locked */ +}; + +struct snd_firewire_event_dice_notification { + unsigned int type; + unsigned int notification; /* DICE-specific bits */ +}; + +union snd_firewire_event { + struct snd_firewire_event_common common; + struct snd_firewire_event_lock_status lock_status; + struct snd_firewire_event_dice_notification dice_notification; +}; + + +#define SNDRV_FIREWIRE_IOCTL_GET_INFO _IOR('H', 0xf8, struct snd_firewire_get_info) +#define SNDRV_FIREWIRE_IOCTL_LOCK _IO('H', 0xf9) +#define SNDRV_FIREWIRE_IOCTL_UNLOCK _IO('H', 0xfa) + +#define SNDRV_FIREWIRE_TYPE_DICE 1 +/* Fireworks, AV/C, RME, MOTU, ... */ + +struct snd_firewire_get_info { + unsigned int type; /* SNDRV_FIREWIRE_TYPE_xxx */ + unsigned int card; /* same as fw_cdev_get_info.card */ + unsigned char guid[8]; + char device_name[16]; /* device node in /dev */ +}; + +/* + * SNDRV_FIREWIRE_IOCTL_LOCK prevents the driver from streaming. + * Returns -EBUSY if the driver is already streaming. + */ + +#endif diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index ea063e1f8722..915330989412 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -11,6 +11,19 @@ config SND_FIREWIRE_LIB tristate depends on SND_PCM +config SND_DICE + tristate "DICE devices (EXPERIMENTAL)" + select SND_HWDEP + select SND_PCM + select SND_FIREWIRE_LIB + help + Say Y here to include support for many FireWire audio interfaces + based on the DICE chip family (DICE-II/Jr/Mini) from TC Applied + Technologies. + + To compile this driver as a module, choose M here: the module + will be called snd-dice. + config SND_FIREWIRE_SPEAKERS tristate "FireWire speakers" select SND_PCM diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 460179df5bb5..509955061d30 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,10 +1,12 @@ snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp.o +snd-dice-objs := dice.o snd-firewire-speakers-objs := speakers.o snd-isight-objs := isight.o snd-scs1x-objs := scs1x.o obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o +obj-$(CONFIG_SND_DICE) += snd-dice.o obj-$(CONFIG_SND_FIREWIRE_SPEAKERS) += snd-firewire-speakers.o obj-$(CONFIG_SND_ISIGHT) += snd-isight.o obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c new file mode 100644 index 000000000000..ac71b2b94eec --- /dev/null +++ b/sound/firewire/dice.c @@ -0,0 +1,1008 @@ +/* + * TC Applied Technologies Digital Interface Communications Engine driver + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "amdtp.h" +#include "iso-resources.h" +#include "lib.h" + +#define DICE_PRIVATE_SPACE 0xffffe0000000uLL + +/* offset from DICE_PRIVATE_SPACE; offsets and sizes in quadlets */ +#define DICE_GLOBAL_OFFSET 0x00 +#define DICE_GLOBAL_SIZE 0x04 +#define DICE_TX_OFFSET 0x08 +#define DICE_TX_SIZE 0x0c +#define DICE_RX_OFFSET 0x10 +#define DICE_RX_SIZE 0x14 + +/* pointed to by DICE_GLOBAL_OFFSET */ +#define GLOBAL_OWNER 0x000 +#define OWNER_NO_OWNER 0xffff000000000000uLL +#define OWNER_NODE_SHIFT 48 +#define GLOBAL_NOTIFICATION 0x008 +#define NOTIFY_RX_CFG_CHG 0x00000001 +#define NOTIFY_TX_CFG_CHG 0x00000002 +#define NOTIFY_DUP_ISOC 0x00000004 +#define NOTIFY_BW_ERR 0x00000008 +#define NOTIFY_LOCK_CHG 0x00000010 +#define NOTIFY_CLOCK_ACCEPTED 0x00000020 +#define NOTIFY_INTERFACE_CHG 0x00000040 +#define NOTIFY_MESSAGE 0x00100000 +#define GLOBAL_NICK_NAME 0x00c +#define NICK_NAME_SIZE 64 +#define GLOBAL_CLOCK_SELECT 0x04c +#define CLOCK_SOURCE_MASK 0x000000ff +#define CLOCK_SOURCE_AES1 0x00000000 +#define CLOCK_SOURCE_AES2 0x00000001 +#define CLOCK_SOURCE_AES3 0x00000002 +#define CLOCK_SOURCE_AES4 0x00000003 +#define CLOCK_SOURCE_AES_ANY 0x00000004 +#define CLOCK_SOURCE_ADAT 0x00000005 +#define CLOCK_SOURCE_TDIF 0x00000006 +#define CLOCK_SOURCE_WC 0x00000007 +#define CLOCK_SOURCE_ARX1 0x00000008 +#define CLOCK_SOURCE_ARX2 0x00000009 +#define CLOCK_SOURCE_ARX3 0x0000000a +#define CLOCK_SOURCE_ARX4 0x0000000b +#define CLOCK_SOURCE_INTERNAL 0x0000000c +#define CLOCK_RATE_MASK 0x0000ff00 +#define CLOCK_RATE_32000 0x00000000 +#define CLOCK_RATE_44100 0x00000100 +#define CLOCK_RATE_48000 0x00000200 +#define CLOCK_RATE_88200 0x00000300 +#define CLOCK_RATE_96000 0x00000400 +#define CLOCK_RATE_176400 0x00000500 +#define CLOCK_RATE_192000 0x00000600 +#define CLOCK_RATE_ANY_LOW 0x00000700 +#define CLOCK_RATE_ANY_MID 0x00000800 +#define CLOCK_RATE_ANY_HIGH 0x00000900 +#define CLOCK_RATE_NONE 0x00000a00 +#define GLOBAL_ENABLE 0x050 +#define ENABLE 0x00000001 +#define GLOBAL_STATUS 0x054 +#define STATUS_SOURCE_LOCKED 0x00000001 +#define STATUS_RATE_CONFLICT 0x00000002 +#define STATUS_NOMINAL_RATE_MASK 0x0000ff00 +#define GLOBAL_EXTENDED_STATUS 0x058 +#define EXT_STATUS_AES1_LOCKED 0x00000001 +#define EXT_STATUS_AES2_LOCKED 0x00000002 +#define EXT_STATUS_AES3_LOCKED 0x00000004 +#define EXT_STATUS_AES4_LOCKED 0x00000008 +#define EXT_STATUS_ADAT_LOCKED 0x00000010 +#define EXT_STATUS_TDIF_LOCKED 0x00000020 +#define EXT_STATUS_ARX1_LOCKED 0x00000040 +#define EXT_STATUS_ARX2_LOCKED 0x00000080 +#define EXT_STATUS_ARX3_LOCKED 0x00000100 +#define EXT_STATUS_ARX4_LOCKED 0x00000200 +#define EXT_STATUS_WC_LOCKED 0x00000400 +#define EXT_STATUS_AES1_SLIP 0x00010000 +#define EXT_STATUS_AES2_SLIP 0x00020000 +#define EXT_STATUS_AES3_SLIP 0x00040000 +#define EXT_STATUS_AES4_SLIP 0x00080000 +#define EXT_STATUS_ADAT_SLIP 0x00100000 +#define EXT_STATUS_TDIF_SLIP 0x00200000 +#define EXT_STATUS_ARX1_SLIP 0x00400000 +#define EXT_STATUS_ARX2_SLIP 0x00800000 +#define EXT_STATUS_ARX3_SLIP 0x01000000 +#define EXT_STATUS_ARX4_SLIP 0x02000000 +#define EXT_STATUS_WC_SLIP 0x04000000 +#define GLOBAL_SAMPLE_RATE 0x05c +#define GLOBAL_VERSION 0x060 +#define GLOBAL_CLOCK_CAPABILITIES 0x064 +#define CLOCK_CAP_RATE_32000 0x00000001 +#define CLOCK_CAP_RATE_44100 0x00000002 +#define CLOCK_CAP_RATE_48000 0x00000004 +#define CLOCK_CAP_RATE_88200 0x00000008 +#define CLOCK_CAP_RATE_96000 0x00000010 +#define CLOCK_CAP_RATE_176400 0x00000020 +#define CLOCK_CAP_RATE_192000 0x00000040 +#define CLOCK_CAP_SOURCE_AES1 0x00010000 +#define CLOCK_CAP_SOURCE_AES2 0x00020000 +#define CLOCK_CAP_SOURCE_AES3 0x00040000 +#define CLOCK_CAP_SOURCE_AES4 0x00080000 +#define CLOCK_CAP_SOURCE_AES_ANY 0x00100000 +#define CLOCK_CAP_SOURCE_ADAT 0x00200000 +#define CLOCK_CAP_SOURCE_TDIF 0x00400000 +#define CLOCK_CAP_SOURCE_WC 0x00800000 +#define CLOCK_CAP_SOURCE_ARX1 0x01000000 +#define CLOCK_CAP_SOURCE_ARX2 0x02000000 +#define CLOCK_CAP_SOURCE_ARX3 0x04000000 +#define CLOCK_CAP_SOURCE_ARX4 0x08000000 +#define CLOCK_CAP_SOURCE_INTERNAL 0x10000000 +#define GLOBAL_CLOCK_SOURCE_NAMES 0x068 +#define CLOCK_SOURCE_NAMES_SIZE 256 + +/* pointed to by DICE_TX_OFFSET */ +#define TX_NUMBER 0x000 +#define TX_SIZE 0x004 +/* repeated TX_NUMBER times, offset by TX_SIZE quadlets */ +#define TX_ISOCHRONOUS 0x008 +#define TX_NUMBER_AUDIO 0x00c +#define TX_NUMBER_MIDI 0x010 +#define TX_SPEED 0x014 +#define TX_NAMES 0x018 +#define TX_NAMES_SIZE 256 +#define TX_AC3_CAPABILITIES 0x118 +#define TX_AC3_ENABLE 0x11c + +/* pointed to by DICE_RX_OFFSET */ +#define RX_NUMBER 0x000 +#define RX_SIZE 0x004 +/* repeated RX_NUMBER times, offset by RX_SIZE quadlets */ +#define RX_ISOCHRONOUS 0x008 +#define RX_SEQ_START 0x00c +#define RX_NUMBER_AUDIO 0x010 +#define RX_NUMBER_MIDI 0x014 +#define RX_NAMES 0x018 +#define RX_NAMES_SIZE 256 +#define RX_AC3_CAPABILITIES 0x118 +#define RX_AC3_ENABLE 0x11c + + +#define FIRMWARE_LOAD_SPACE 0xffffe0100000uLL + +/* offset from FIRMWARE_LOAD_SPACE */ +#define FIRMWARE_VERSION 0x000 +#define FIRMWARE_OPCODE 0x004 +#define OPCODE_MASK 0x00000fff +#define OPCODE_GET_IMAGE_DESC 0x00000000 +#define OPCODE_DELETE_IMAGE 0x00000001 +#define OPCODE_CREATE_IMAGE 0x00000002 +#define OPCODE_UPLOAD 0x00000003 +#define OPCODE_UPLOAD_STAT 0x00000004 +#define OPCODE_RESET_IMAGE 0x00000005 +#define OPCODE_TEST_ACTION 0x00000006 +#define OPCODE_GET_RUNNING_IMAGE_VINFO 0x0000000a +#define OPCODE_EXECUTE 0x80000000 +#define FIRMWARE_RETURN_STATUS 0x008 +#define FIRMWARE_PROGRESS 0x00c +#define PROGRESS_CURR_MASK 0x00000fff +#define PROGRESS_MAX_MASK 0x00fff000 +#define PROGRESS_TOUT_MASK 0x0f000000 +#define PROGRESS_FLAG 0x80000000 +#define FIRMWARE_CAPABILITIES 0x010 +#define FL_CAP_AUTOERASE 0x00000001 +#define FL_CAP_PROGRESS 0x00000002 +#define FIRMWARE_DATA 0x02c +#define TEST_CMD_POKE 0x00000001 +#define TEST_CMD_PEEK 0x00000002 +#define CMD_GET_AVS_CNT 0x00000003 +#define CMD_CLR_AVS_CNT 0x00000004 +#define CMD_SET_MODE 0x00000005 +#define CMD_SET_MIDIBP 0x00000006 +#define CMD_GET_AVSPHASE 0x00000007 +#define CMD_ENABLE_BNC_SYNC 0x00000008 +#define CMD_PULSE_BNC_SYNC 0x00000009 +#define CMD_EMUL_SLOW_CMD 0x0000000a +#define FIRMWARE_TEST_DELAY 0xfd8 +#define FIRMWARE_TEST_BUF 0xfdc + + +/* EAP */ +#define EAP_PRIVATE_SPACE 0xffffe0200000uLL + +#define EAP_CAPABILITY_OFFSET 0x000 +#define EAP_CAPABILITY_SIZE 0x004 +/* ... */ + +#define EAP_ROUTER_CAPS 0x000 +#define ROUTER_EXPOSED 0x00000001 +#define ROUTER_READ_ONLY 0x00000002 +#define ROUTER_FLASH 0x00000004 +#define MAX_ROUTES_MASK 0xffff0000 +#define EAP_MIXER_CAPS 0x004 +#define MIXER_EXPOSED 0x00000001 +#define MIXER_READ_ONLY 0x00000002 +#define MIXER_FLASH 0x00000004 +#define MIXER_IN_DEV_MASK 0x000000f0 +#define MIXER_OUT_DEV_MASK 0x00000f00 +#define MIXER_INPUTS_MASK 0x00ff0000 +#define MIXER_OUTPUTS_MASK 0xff000000 +#define EAP_GENERAL_CAPS 0x008 +#define GENERAL_STREAM_CONFIG 0x00000001 +#define GENERAL_FLASH 0x00000002 +#define GENERAL_PEAK 0x00000004 +#define GENERAL_MAX_TX_STREAMS_MASK 0x000000f0 +#define GENERAL_MAX_RX_STREAMS_MASK 0x00000f00 +#define GENERAL_STREAM_CONFIG_FLASH 0x00001000 +#define GENERAL_CHIP_MASK 0x00ff0000 +#define GENERAL_CHIP_DICE_II 0x00000000 +#define GENERAL_CHIP_DICE_MINI 0x00010000 +#define GENERAL_CHIP_DICE_JR 0x00020000 + + +struct dice { + struct snd_card *card; + struct fw_unit *unit; + struct mutex mutex; + unsigned int global_offset; + unsigned int rx_offset; + struct fw_address_handler notification_handler; + int owner_generation; + bool global_enabled; + bool stream_running; + struct snd_pcm_substream *pcm; + struct fw_iso_resources resources; + struct amdtp_out_stream stream; +}; + +MODULE_DESCRIPTION("DICE driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); + +static inline u64 global_address(struct dice *dice, unsigned int offset) +{ + return DICE_PRIVATE_SPACE + dice->global_offset + offset; +} + +// TODO: rx index +static inline u64 rx_address(struct dice *dice, unsigned int offset) +{ + return DICE_PRIVATE_SPACE + dice->rx_offset + offset; +} + +static int dice_owner_set(struct dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + int rcode, err, errors = 0; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + + for (;;) { + buffer[0] = cpu_to_be64(OWNER_NO_OWNER); + buffer[1] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + + dice->owner_generation = device->generation; + smp_rmb(); /* node_id vs. generation */ + rcode = fw_run_transaction(device->card, + TCODE_LOCK_COMPARE_SWAP, + device->node_id, + dice->owner_generation, + device->max_speed, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8); + + if (rcode == RCODE_COMPLETE) { + if (buffer[0] == cpu_to_be64(OWNER_NO_OWNER)) { + err = 0; + } else { + dev_err(&dice->unit->device, + "device is already in use\n"); + err = -EBUSY; + } + break; + } + if (rcode_is_permanent_error(rcode) || ++errors >= 3) { + dev_err(&dice->unit->device, + "setting device owner failed: %s\n", + fw_rcode_string(rcode)); + err = -EIO; + break; + } + msleep(20); + } + + kfree(buffer); + + return err; +} + +static int dice_owner_update(struct dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + int rcode, err, errors = 0; + + if (dice->owner_generation == -1) + return 0; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + + for (;;) { + buffer[0] = cpu_to_be64(OWNER_NO_OWNER); + buffer[1] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + + dice->owner_generation = device->generation; + smp_rmb(); /* node_id vs. generation */ + rcode = fw_run_transaction(device->card, + TCODE_LOCK_COMPARE_SWAP, + device->node_id, + dice->owner_generation, + device->max_speed, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8); + + if (rcode == RCODE_COMPLETE) { + if (buffer[0] == cpu_to_be64(OWNER_NO_OWNER)) { + err = 0; + } else { + dev_err(&dice->unit->device, + "device is already in use\n"); + err = -EBUSY; + } + break; + } + if (rcode == RCODE_GENERATION) { + err = 0; /* try again later */ + break; + } + if (rcode_is_permanent_error(rcode) || ++errors >= 3) { + dev_err(&dice->unit->device, + "setting device owner failed: %s\n", + fw_rcode_string(rcode)); + err = -EIO; + break; + } + msleep(20); + } + + kfree(buffer); + + if (err < 0) + dice->owner_generation = -1; + + return err; +} + +static void dice_owner_clear(struct dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + int rcode, errors = 0; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (!buffer) + return; + + for (;;) { + buffer[0] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + buffer[1] = cpu_to_be64(OWNER_NO_OWNER); + + rcode = fw_run_transaction(device->card, + TCODE_LOCK_COMPARE_SWAP, + device->node_id, + dice->owner_generation, + device->max_speed, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8); + + if (rcode == RCODE_COMPLETE) + break; + if (rcode == RCODE_GENERATION) + break; + if (rcode_is_permanent_error(rcode) || ++errors >= 3) { + dev_err(&dice->unit->device, + "clearing device owner failed: %s\n", + fw_rcode_string(rcode)); + break; + } + msleep(20); + } + + kfree(buffer); + + dice->owner_generation = -1; +} + +static int dice_enable_set(struct dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be32 value; + int rcode, err, errors = 0; + + value = cpu_to_be32(ENABLE); + for (;;) { + rcode = fw_run_transaction(device->card, + TCODE_WRITE_QUADLET_REQUEST, + device->node_id, + dice->owner_generation, + device->max_speed, + global_address(dice, GLOBAL_ENABLE), + &value, 4); + if (rcode == RCODE_COMPLETE) { + dice->global_enabled = true; + err = 0; + break; + } + if (rcode == RCODE_GENERATION) { + err = -EAGAIN; + break; + } + if (rcode_is_permanent_error(rcode) || ++errors >= 3) { + dev_err(&dice->unit->device, + "device enabling failed: %s\n", + fw_rcode_string(rcode)); + err = -EIO; + break; + } + msleep(20); + } + + return err; +} + +static void dice_enable_clear(struct dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be32 value; + int rcode, errors = 0; + + value = 0; + for (;;) { + rcode = fw_run_transaction(device->card, + TCODE_WRITE_QUADLET_REQUEST, + device->node_id, + dice->owner_generation, + device->max_speed, + global_address(dice, GLOBAL_ENABLE), + &value, 4); + if (rcode == RCODE_COMPLETE || + rcode == RCODE_GENERATION) + break; + if (rcode_is_permanent_error(rcode) || ++errors >= 3) { + dev_err(&dice->unit->device, + "device disabling failed: %s\n", + fw_rcode_string(rcode)); + break; + } + msleep(20); + } + dice->global_enabled = false; +} + +static void dice_notification(struct fw_card *card, struct fw_request *request, + int tcode, int destination, int source, + int generation, unsigned long long offset, + void *data, size_t length, void *callback_data) +{ + struct dice *dice = callback_data; + + if (tcode != TCODE_WRITE_QUADLET_REQUEST) { + fw_send_response(card, request, RCODE_TYPE_ERROR); + return; + } + if ((offset & 3) != 0) { + fw_send_response(card, request, RCODE_ADDRESS_ERROR); + return; + } + dev_info(&dice->unit->device, + "notification: %08x\n", be32_to_cpup(data)); + fw_send_response(card, request, RCODE_COMPLETE); +} + +static int dice_open(struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = AMDTP_OUT_PCM_FORMAT_BITS, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .buffer_bytes_max = 16 * 1024 * 1024, + .period_bytes_min = 1, + .period_bytes_max = UINT_MAX, + .periods_min = 1, + .periods_max = UINT_MAX, + }; + struct dice *dice = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + __be32 number_audio, number_midi; + int err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + rx_address(dice, RX_NUMBER_AUDIO), + &number_audio, 4); + if (err < 0) + return err; + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + rx_address(dice, RX_NUMBER_MIDI), + &number_midi, 4); + if (err < 0) + return err; + + runtime->hw = hardware; + runtime->hw.channels_min = be32_to_cpu(number_audio); + runtime->hw.channels_max = be32_to_cpu(number_audio); + + amdtp_out_stream_set_rate(&dice->stream, 44100); + amdtp_out_stream_set_pcm(&dice->stream, be32_to_cpu(number_audio)); + amdtp_out_stream_set_midi(&dice->stream, be32_to_cpu(number_midi)); + + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 5000, 8192000); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + if (err < 0) + return err; + + return 0; +} + +static int dice_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static void dice_stop_stream(struct dice *dice) +{ + __be32 channel; + + if (dice->stream_running) { + dice_enable_clear(dice); + + amdtp_out_stream_stop(&dice->stream); + + channel = cpu_to_be32((u32)-1); + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), + &channel, 4); + + fw_iso_resources_free(&dice->resources); + + dice->stream_running = false; + } +} + +static int dice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct dice *dice = substream->private_data; + int err; + + mutex_lock(&dice->mutex); + dice_stop_stream(dice); + mutex_unlock(&dice->mutex); + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + goto error; + + amdtp_out_stream_set_pcm_format(&dice->stream, + params_format(hw_params)); + + return 0; + +error: + return err; +} + +static int dice_hw_free(struct snd_pcm_substream *substream) +{ + struct dice *dice = substream->private_data; + + mutex_lock(&dice->mutex); + dice_stop_stream(dice); + mutex_unlock(&dice->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int dice_prepare(struct snd_pcm_substream *substream) +{ + struct dice *dice = substream->private_data; + struct fw_device *device = fw_parent_device(dice->unit); + __be32 channel; + int err; + + mutex_lock(&dice->mutex); + + if (amdtp_out_streaming_error(&dice->stream)) + dice_stop_stream(dice); + + if (!dice->stream_running) { + err = fw_iso_resources_allocate(&dice->resources, + amdtp_out_stream_get_max_payload(&dice->stream), + device->max_speed); + if (err < 0) + goto error; + + //TODO: RX_SEQ_START + channel = cpu_to_be32(dice->resources.channel); + err = snd_fw_transaction(dice->unit, + TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), + &channel, 4); + if (err < 0) + goto err_resources; + + err = amdtp_out_stream_start(&dice->stream, + dice->resources.channel, + device->max_speed); + if (err < 0) + goto err_resources; + + err = dice_enable_set(dice); + if (err < 0) + goto err_stream; + + dice->stream_running = true; + } + + mutex_unlock(&dice->mutex); + + amdtp_out_stream_pcm_prepare(&dice->stream); + + return 0; + +err_stream: + amdtp_out_stream_stop(&dice->stream); +err_resources: + fw_iso_resources_free(&dice->resources); +error: + mutex_unlock(&dice->mutex); + + return err; +} + +static int dice_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct dice *dice = substream->private_data; + struct snd_pcm_substream *pcm; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pcm = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pcm = NULL; + break; + default: + return -EINVAL; + } + amdtp_out_stream_pcm_trigger(&dice->stream, pcm); + + return 0; +} + +static snd_pcm_uframes_t dice_pointer(struct snd_pcm_substream *substream) +{ + struct dice *dice = substream->private_data; + + return amdtp_out_stream_pcm_pointer(&dice->stream); +} + +static int dice_create_pcm(struct dice *dice) +{ + static struct snd_pcm_ops ops = { + .open = dice_open, + .close = dice_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dice_hw_params, + .hw_free = dice_hw_free, + .prepare = dice_prepare, + .trigger = dice_trigger, + .pointer = dice_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + __be32 clock; + struct snd_pcm *pcm; + int err; + + clock = cpu_to_be32(CLOCK_SOURCE_ARX1 | CLOCK_RATE_44100); + err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &clock, 4); + if (err < 0) + return err; + + err = snd_pcm_new(dice->card, "DICE", 0, 1, 0, &pcm); + if (err < 0) + return err; + pcm->private_data = dice; + strcpy(pcm->name, dice->card->shortname); + dice->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + dice->pcm->ops = &ops; + + return 0; +} + +// TODO: implement these + +static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, + long count, loff_t *offset) +{ + return -EIO; +} + +static int dice_hwdep_open(struct snd_hwdep *hwdep, struct file *file) +{ + return -EIO; +} + +static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + return 0; +} + +static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + return POLLERR | POLLHUP; +} + +static int dice_hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return -EIO; +} + +static int dice_create_hwdep(struct dice *dice) +{ + static const struct snd_hwdep_ops ops = { + .read = dice_hwdep_read, + .open = dice_hwdep_open, + .release = dice_hwdep_release, + .poll = dice_hwdep_poll, + .ioctl = dice_hwdep_ioctl, + .ioctl_compat = dice_hwdep_ioctl, + }; + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(dice->card, "DICE", 0, &hwdep); + if (err < 0) + return err; + strcpy(hwdep->name, "DICE"); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_DICE; + hwdep->ops = ops; + hwdep->private_data = dice; + hwdep->exclusive = true; + + return 0; +} + +static void dice_card_free(struct snd_card *card) +{ + struct dice *dice = card->private_data; + + amdtp_out_stream_destroy(&dice->stream); + fw_core_remove_address_handler(&dice->notification_handler); + mutex_destroy(&dice->mutex); +} + +static int dice_init_offsets(struct dice *dice) +{ + __be32 pointers[6]; + unsigned int global_size, rx_size; + int err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE, &pointers, 6 * 4); + if (err < 0) + return err; + + dice->global_offset = be32_to_cpu(pointers[0]) * 4; + global_size = be32_to_cpu(pointers[1]); + dice->rx_offset = be32_to_cpu(pointers[4]) * 4; + rx_size = be32_to_cpu(pointers[5]); + + /* some sanity checks to ensure that we actually have a DICE */ + if (dice->global_offset < 10 * 4 || global_size < 0x168 / 4 || + dice->rx_offset < 10 * 4 || rx_size < 0x120 / 4) { + dev_err(&dice->unit->device, "invalid register pointers\n"); + return -ENXIO; + } + + return 0; +} + +static void dice_card_strings(struct dice *dice) +{ + struct snd_card *card = dice->card; + struct fw_device *dev = fw_parent_device(dice->unit); + char vendor[32], model[32]; + unsigned int i; + int err; + + strcpy(card->driver, "DICE"); + + strcpy(card->shortname, "DICE"); + BUILD_BUG_ON(NICK_NAME_SIZE < sizeof(card->shortname)); + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + global_address(dice, GLOBAL_NICK_NAME), + card->shortname, sizeof(card->shortname)); + if (err >= 0) { + /* DICE strings are returned in "always-wrong" endianness */ + BUILD_BUG_ON(sizeof(card->shortname) % 4 != 0); + for (i = 0; i < sizeof(card->shortname); i += 4) + swab32s((u32 *)&card->shortname[i]); + card->shortname[sizeof(card->shortname) - 1] = '\0'; + } + + strcpy(vendor, "?"); + fw_csr_string(dev->config_rom + 5, CSR_VENDOR, vendor, sizeof(vendor)); + strcpy(model, "?"); + fw_csr_string(dice->unit->directory, CSR_MODEL, model, sizeof(model)); + snprintf(card->longname, sizeof(card->longname), + "%s %s, GUID %08x%08x at %s, S%d", + vendor, model, dev->config_rom[3], dev->config_rom[4], + dev_name(&dice->unit->device), 100 << dev->max_speed); + + strcpy(card->mixername, "DICE"); +} + +static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) +{ + struct snd_card *card; + struct dice *dice; + int err; + + err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*dice), &card); + if (err < 0) + return err; + snd_card_set_dev(card, &unit->device); + + dice = card->private_data; + dice->card = card; + mutex_init(&dice->mutex); + dice->unit = unit; + + err = dice_init_offsets(dice); + if (err < 0) + goto err_mutex; + + dice->notification_handler.length = 4; + dice->notification_handler.address_callback = dice_notification; + dice->notification_handler.callback_data = dice; + err = fw_core_add_address_handler(&dice->notification_handler, + &fw_high_memory_region); + if (err < 0) + goto err_mutex; + + err = fw_iso_resources_init(&dice->resources, unit); + if (err < 0) + goto err_notification_handler; + dice->resources.channels_mask = 0x00000000ffffffffuLL; + + err = amdtp_out_stream_init(&dice->stream, unit, CIP_NONBLOCKING); + if (err < 0) + goto err_resources; + + err = dice_owner_set(dice); + if (err < 0) + goto err_stream; + + card->private_free = dice_card_free; + + dice_card_strings(dice); + + err = dice_create_pcm(dice); + if (err < 0) + goto error; + + err = dice_create_hwdep(dice); + if (err < 0) + goto error; + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(&unit->device, dice); + + return 0; + +err_stream: + amdtp_out_stream_destroy(&dice->stream); +err_resources: + fw_iso_resources_destroy(&dice->resources); +err_notification_handler: + fw_core_remove_address_handler(&dice->notification_handler); +err_mutex: + mutex_destroy(&dice->mutex); +error: + snd_card_free(card); + return err; +} + +static void dice_remove(struct fw_unit *unit) +{ + struct dice *dice = dev_get_drvdata(&unit->device); + + snd_card_disconnect(dice->card); + + mutex_lock(&dice->mutex); + amdtp_out_stream_pcm_abort(&dice->stream); + dice_stop_stream(dice); + dice_owner_clear(dice); + mutex_unlock(&dice->mutex); + + snd_card_free_when_closed(dice->card); +} + +static void dice_bus_reset(struct fw_unit *unit) +{ + struct dice *dice = dev_get_drvdata(&unit->device); + + mutex_lock(&dice->mutex); + /* + * XXX is the following true? + * On a bus reset, the DICE firmware disables streaming and then goes + * off contemplating its own navel for hundreds of milliseconds before + * it can react to any of our attempts to reenable streaming. This + * means that we lose synchronization anyway, so we force our streams + * to stop so that the application can restart them in an orderly + * manner. + */ + dice_owner_update(dice); + amdtp_out_stream_pcm_abort(&dice->stream); + dice_stop_stream(dice); + mutex_unlock(&dice->mutex); +} + +#define TC_OUI 0x000166 +#define DICE_INTERFACE 0x000001 + +static const struct ieee1394_device_id dice_id_table[] = { + { + .match_flags = IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .specifier_id = TC_OUI, + .version = DICE_INTERFACE, + }, + { } +}; +MODULE_DEVICE_TABLE(ieee1394, dice_id_table); + +static struct fw_driver dice_driver = { + .driver = { + .owner = THIS_MODULE, + .name = KBUILD_MODNAME, + .bus = &fw_bus_type, + }, + .probe = dice_probe, + .update = dice_bus_reset, + .remove = dice_remove, + .id_table = dice_id_table, +}; + +static int __init alsa_dice_init(void) +{ + return driver_register(&dice_driver.driver); +} + +static void __exit alsa_dice_exit(void) +{ + driver_unregister(&dice_driver.driver); +} + +module_init(alsa_dice_init); +module_exit(alsa_dice_exit); -- cgit From beb02cddd64b56081951de9048952f0fa1ff545f Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Thu, 17 Oct 2013 14:01:35 +0400 Subject: ALSA: pxa: slightly refactor reset handling PXA25x also shows some problems when using interrupts during reset handling. Thus do not use interrupts on all pxa kinds (to detect codec ready state). Instead use a common mdelay-loop on all platforms to detect codecs becoming ready. Signed-off-by: Dmitry Eremin-Solenikov Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 27 ++++++++++----------------- 1 file changed, 10 insertions(+), 17 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index e6f4633b8dd5..99a466822a7d 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -117,8 +117,7 @@ static inline void pxa_ac97_warm_pxa25x(void) { gsr_bits = 0; - GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); + GCR |= GCR_WARM_RST; } static inline void pxa_ac97_cold_pxa25x(void) @@ -129,8 +128,6 @@ static inline void pxa_ac97_cold_pxa25x(void) gsr_bits = 0; GCR = GCR_COLD_RST; - GCR |= GCR_CDONE_IE|GCR_SDONE_IE; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); } #endif @@ -149,8 +146,6 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { - unsigned int timeout; - GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -161,29 +156,20 @@ static inline void pxa_ac97_cold_pxa27x(void) udelay(5); clk_disable(ac97conf_clk); GCR = GCR_COLD_RST | GCR_WARM_RST; - timeout = 100; /* wait for the codec-ready bit to be set */ - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); } #endif #ifdef CONFIG_PXA3xx static inline void pxa_ac97_warm_pxa3xx(void) { - int timeout = 100; - gsr_bits = 0; /* Can't use interrupts */ GCR |= GCR_WARM_RST; - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); } static inline void pxa_ac97_cold_pxa3xx(void) { - int timeout = 1000; - /* Hold CLKBPB for 100us */ GCR = 0; GCR = GCR_CLKBPB; @@ -199,14 +185,13 @@ static inline void pxa_ac97_cold_pxa3xx(void) GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR = GCR_WARM_RST | GCR_COLD_RST; - while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(10); } #endif bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { unsigned long gsr; + unsigned int timeout = 100; #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) @@ -224,6 +209,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) else #endif BUG(); + + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); + gsr = GSR | gsr_bits; if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", @@ -239,6 +228,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { unsigned long gsr; + unsigned int timeout = 1000; #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) @@ -257,6 +247,9 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) #endif BUG(); + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); + gsr = GSR | gsr_bits; if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", -- cgit From f62aa9b6c900a9aaf302c1f9ac1883c143afd832 Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Thu, 17 Oct 2013 14:01:36 +0400 Subject: ALSA: ASoC: pxa: fix pxa2xx-ac97 DAI initialization order After recent changes to codec/DAI initialization order changes, codec driver (wm9712 in my case) tries to access codec prior to pxa2xx_ac97_hw_probe() being called (because DAIs are probed after all codecs are probed). Move hw-related probe/remove/suspend/resume functions to pxa2xx-ac97 driver level, instead of DAI level. Signed-off-by: Dmitry Eremin-Solenikov Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-ac97.c | 56 ++++++++++++++++++++------------------------- 1 file changed, 25 insertions(+), 31 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f1059d999de6..ae956e3f4b9d 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -89,33 +89,6 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; -#ifdef CONFIG_PM -static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_suspend(); -} - -static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_resume(); -} - -#else -#define pxa2xx_ac97_suspend NULL -#define pxa2xx_ac97_resume NULL -#endif - -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -185,10 +158,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, - .suspend = pxa2xx_ac97_suspend, - .resume = pxa2xx_ac97_resume, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -246,6 +215,12 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return -ENXIO; } + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret) { + dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret); + return ret; + } + ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); if (ret != 0) return ret; @@ -262,15 +237,34 @@ static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); + pxa2xx_ac97_hw_remove(pdev); return 0; } +#ifdef CONFIG_PM_SLEEP +static int pxa2xx_ac97_dev_suspend(struct device *dev) +{ + return pxa2xx_ac97_hw_suspend(); +} + +static int pxa2xx_ac97_dev_resume(struct device *dev) +{ + return pxa2xx_ac97_hw_resume(); +} + +static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, + pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume); +#endif + static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_dev_probe, .remove = pxa2xx_ac97_dev_remove, .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, +#ifdef CONFIG_PM_SLEEP + .pm = &pxa2xx_ac97_pm_ops, +#endif }, }; -- cgit From 7db1698f728e1176cc7869f22565e3faa8ec2b72 Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Thu, 17 Oct 2013 14:01:37 +0400 Subject: ALSA: ASoC: pxa: add asoc pm callbacks to pxa audio drivers After convertion to snd_soc_register_card, platform driver should reference snd_soc_pm_ops callbacks to properly suspend/resume sound hardware. This was missed during conversion of PXA sound devices. Signed-off-by: Dmitry Eremin-Solenikov Signed-off-by: Mark Brown --- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 1 + sound/soc/pxa/e750_wm9705.c | 1 + sound/soc/pxa/e800_wm9712.c | 1 + sound/soc/pxa/imote2.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/poodle.c | 1 + sound/soc/pxa/tosa.c | 1 + sound/soc/pxa/ttc-dkb.c | 1 + 11 files changed, 11 insertions(+) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 5b7d969f89a9..08acdc236bf8 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -163,6 +163,7 @@ static struct platform_driver mmp_driver = { .driver = { .name = "brownstone-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = brownstone_probe, .remove = brownstone_remove, diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index f4cce1e80112..1853d41034bf 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -329,6 +329,7 @@ static struct platform_driver corgi_driver = { .driver = { .name = "corgi-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = corgi_probe, .remove = corgi_remove, diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 70d799b13f0d..44b5c09d296b 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -178,6 +178,7 @@ static struct platform_driver e740_driver = { .driver = { .name = "e740-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = e740_probe, .remove = e740_remove, diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index f94d2ab51351..c34e447eb991 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -160,6 +160,7 @@ static struct platform_driver e750_driver = { .driver = { .name = "e750-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = e750_probe, .remove = e750_remove, diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 8768a640dd71..3137f800b43f 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -150,6 +150,7 @@ static struct platform_driver e800_driver = { .driver = { .name = "e800-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = e800_probe, .remove = e800_remove, diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index eef1f7b7b38e..fd2f4eda1fd3 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -91,6 +91,7 @@ static struct platform_driver imote2_driver = { .driver = { .name = "imote2-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = imote2_probe, .remove = imote2_remove, diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index bbea7780eac6..160c5245448f 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -215,6 +215,7 @@ static struct platform_driver mioa701_wm9713_driver = { .driver = { .name = "mioa701-wm9713", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index e1ffcdd9a649..3284c4b901cb 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -181,6 +181,7 @@ static struct platform_driver palm27x_wm9712_driver = { .driver = { .name = "palm27x-asoc", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index fafe46355c31..c93e138d8dc3 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -303,6 +303,7 @@ static struct platform_driver poodle_driver = { .driver = { .name = "poodle-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = poodle_probe, .remove = poodle_remove, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index a3fe19123f07..1d9c2ed223bc 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -275,6 +275,7 @@ static struct platform_driver tosa_driver = { .driver = { .name = "tosa-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = tosa_probe, .remove = tosa_remove, diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 13c9ee0cb83b..0b535b570622 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -160,6 +160,7 @@ static struct platform_driver ttc_dkb_driver = { .driver = { .name = "ttc-dkb-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = ttc_dkb_probe, .remove = ttc_dkb_remove, -- cgit From 3d8c8bc0250f7cb11f887691b7473b51adcd2bcb Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Oct 2013 11:03:33 -0500 Subject: ASoC: cs42l73: Add platform data support for cs42l73 codec Add support for RST GPIO and Charge Pump Freq in platform data Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- include/sound/cs42l73.h | 22 ++++++++++++++++++++ sound/soc/codecs/cs42l73.c | 51 ++++++++++++++++++++++++++++++---------------- sound/soc/codecs/cs42l73.h | 1 + 3 files changed, 56 insertions(+), 18 deletions(-) create mode 100644 include/sound/cs42l73.h diff --git a/include/sound/cs42l73.h b/include/sound/cs42l73.h new file mode 100644 index 000000000000..f354be4cdc9e --- /dev/null +++ b/include/sound/cs42l73.h @@ -0,0 +1,22 @@ +/* + * linux/sound/cs42l73.h -- Platform data for CS42L73 + * + * Copyright (c) 2012 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS42L73_H +#define __CS42L73_H + +struct cs42l73_platform_data { + /* RST GPIO */ + unsigned int reset_gpio; + unsigned int chgfreq; + int jack_detection; + unsigned int mclk_freq; +}; + +#endif /* __CS42L73_H */ diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3b20c86cdb01..db9d39604d68 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -28,6 +29,7 @@ #include #include #include +#include #include "cs42l73.h" struct sp_config { @@ -35,6 +37,7 @@ struct sp_config { u32 srate; }; struct cs42l73_private { + struct cs42l73_platform_data pdata; struct sp_config config[3]; struct regmap *regmap; u32 sysclk; @@ -310,15 +313,6 @@ static const struct soc_enum ng_delay_enum = SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); -static const char * const charge_pump_freq_text[] = { - "0", "1", "2", "3", "4", - "5", "6", "7", "8", "9", - "10", "11", "12", "13", "14", "15" }; - -static const struct soc_enum charge_pump_enum = - SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, - ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); - static const char * const cs42l73_mono_mix_texts[] = { "Left", "Right", "Mono Mix"}; @@ -511,8 +505,6 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), SOC_ENUM("NG Delay", ng_delay_enum), - SOC_ENUM("Charge Pump Frequency", charge_pump_enum), - SOC_DOUBLE_R_TLV("XSP-IP Volume", CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, attn_tlv), @@ -1367,11 +1359,16 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return ret; } - regcache_cache_only(cs42l73->regmap, true); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + /* Set Charge Pump Frequency */ + if (cs42l73->pdata.chgfreq) + snd_soc_update_bits(codec, CS42L73_CPFCHC, + CS42L73_CHARGEPUMP_MASK, + cs42l73->pdata.chgfreq << 4); + + /* MCLK1 as master clk */ + cs42l73->mclksel = CS42L73_CLKID_MCLK1; cs42l73->mclk = 0; return ret; @@ -1415,6 +1412,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l73_private *cs42l73; + struct cs42l73_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1426,14 +1424,32 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } - i2c_set_clientdata(i2c_client, cs42l73); - cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { ret = PTR_ERR(cs42l73->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); return ret; } + + if (pdata) + cs42l73->pdata = *pdata; + + i2c_set_clientdata(i2c_client, cs42l73); + + if (cs42l73->pdata.reset_gpio) { + ret = gpio_request_one(cs42l73->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L73 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l73->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 1); + } + + regcache_cache_bypass(cs42l73->regmap, true); + /* initialize codec */ ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, ®); devid = (reg & 0xFF) << 12; @@ -1444,7 +1460,6 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, ®); devid |= (reg & 0xF0) >> 4; - if (devid != CS42L73_DEVID) { ret = -ENODEV; dev_err(&i2c_client->dev, @@ -1462,7 +1477,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, dev_info(&i2c_client->dev, "Cirrus Logic CS42L73, Revision: %02X\n", reg & 0xFF); - regcache_cache_only(cs42l73->regmap, true); + regcache_cache_bypass(cs42l73->regmap, false); ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l73, cs42l73_dai, diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index f30a4c4d62e6..4f83d39496a8 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -159,6 +159,7 @@ #define THMOVLD_115C 2 #define THMOVLD_098C 3 +#define CS42L73_CHARGEPUMP_MASK (0xF0) /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ #define SP_3ST (1 << 7) -- cgit From f9ca060680e7c26a88d990ad9370572274b0d54b Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Oct 2013 11:03:34 -0500 Subject: ASoC: cs42l73: Namespace defines for cs42l73 codec Cleanup to namespace the defines for the cs42l73 driver Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 38 ++++++++--------- sound/soc/codecs/cs42l73.h | 104 ++++++++++++++++++++++----------------------- 2 files changed, 70 insertions(+), 72 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index db9d39604d68..89efc3c6aefc 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1047,11 +1047,11 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - mmcc |= MS_MASTER; + mmcc |= CS42L73_MS_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: - mmcc &= ~MS_MASTER; + mmcc &= ~CS42L73_MS_MASTER; break; default: @@ -1063,11 +1063,11 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (format) { case SND_SOC_DAIFMT_I2S: - spc &= ~SPDIF_PCM; + spc &= ~CS42L73_SPDIF_PCM; break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - if (mmcc & MS_MASTER) { + if (mmcc & CS42L73_MS_MASTER) { dev_err(codec->dev, "PCM format in slave mode only\n"); return -EINVAL; @@ -1077,25 +1077,25 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) "PCM format is not supported on ASP port\n"); return -EINVAL; } - spc |= SPDIF_PCM; + spc |= CS42L73_SPDIF_PCM; break; default: return -EINVAL; } - if (spc & SPDIF_PCM) { + if (spc & CS42L73_SPDIF_PCM) { /* Clear PCM mode, clear PCM_BIT_ORDER bit for MSB->LSB */ - spc &= ~(PCM_MODE_MASK | PCM_BIT_ORDER); + spc &= ~(CS42L73_PCM_MODE_MASK | CS42L73_PCM_BIT_ORDER); switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= PCM_MODE0; + spc |= CS42L73_PCM_MODE0; if (inv == SND_SOC_DAIFMT_IB_NF) - spc |= PCM_MODE1; + spc |= CS42L73_PCM_MODE1; break; case SND_SOC_DAIFMT_DSP_A: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= PCM_MODE1; + spc |= CS42L73_PCM_MODE1; break; default: return -EINVAL; @@ -1155,7 +1155,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, int mclk_coeff; int srate = params_rate(params); - if (priv->config[id].mmcc & MS_MASTER) { + if (priv->config[id].mmcc & CS42L73_MS_MASTER) { /* CS42L73 Master */ /* MCLK -> srate */ mclk_coeff = @@ -1174,13 +1174,13 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].spc &= 0xFC; /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */ if (priv->mclk >= 6400000) - priv->config[id].spc |= MCK_SCLK_64FS; + priv->config[id].spc |= CS42L73_MCK_SCLK_64FS; else - priv->config[id].spc |= MCK_SCLK_MCLK; + priv->config[id].spc |= CS42L73_MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; - priv->config[id].spc |= MCK_SCLK_64FS; + priv->config[id].spc |= CS42L73_MCK_SCLK_64FS; } /* Update ASRCs */ priv->config[id].srate = srate; @@ -1200,8 +1200,8 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 0); break; case SND_SOC_BIAS_PREPARE: @@ -1212,11 +1212,11 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(cs42l73->regmap, false); regcache_sync(cs42l73->regmap); } - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1); if (cs42l73->shutdwn_delay > 0) { mdelay(cs42l73->shutdwn_delay); cs42l73->shutdwn_delay = 0; @@ -1225,7 +1225,7 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, * down. */ } - snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 1); break; } codec->dapm.bias_level = level; diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index 4f83d39496a8..45746186a678 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -128,60 +128,60 @@ /* Bitfield Definitions */ /* CS42L73_PWRCTL1 */ -#define PDN_ADCB (1 << 7) -#define PDN_DMICB (1 << 6) -#define PDN_ADCA (1 << 5) -#define PDN_DMICA (1 << 4) -#define PDN_LDO (1 << 2) -#define DISCHG_FILT (1 << 1) -#define PDN (1 << 0) +#define CS42L73_PDN_ADCB (1 << 7) +#define CS42L73_PDN_DMICB (1 << 6) +#define CS42L73_PDN_ADCA (1 << 5) +#define CS42L73_PDN_DMICA (1 << 4) +#define CS42L73_PDN_LDO (1 << 2) +#define CS42L73_DISCHG_FILT (1 << 1) +#define CS42L73_PDN (1 << 0) /* CS42L73_PWRCTL2 */ -#define PDN_MIC2_BIAS (1 << 7) -#define PDN_MIC1_BIAS (1 << 6) -#define PDN_VSP (1 << 4) -#define PDN_ASP_SDOUT (1 << 3) -#define PDN_ASP_SDIN (1 << 2) -#define PDN_XSP_SDOUT (1 << 1) -#define PDN_XSP_SDIN (1 << 0) +#define CS42L73_PDN_MIC2_BIAS (1 << 7) +#define CS42L73_PDN_MIC1_BIAS (1 << 6) +#define CS42L73_PDN_VSP (1 << 4) +#define CS42L73_PDN_ASP_SDOUT (1 << 3) +#define CS42L73_PDN_ASP_SDIN (1 << 2) +#define CS42L73_PDN_XSP_SDOUT (1 << 1) +#define CS42L73_PDN_XSP_SDIN (1 << 0) /* CS42L73_PWRCTL3 */ -#define PDN_THMS (1 << 5) -#define PDN_SPKLO (1 << 4) -#define PDN_EAR (1 << 3) -#define PDN_SPK (1 << 2) -#define PDN_LO (1 << 1) -#define PDN_HP (1 << 0) +#define CS42L73_PDN_THMS (1 << 5) +#define CS42L73_PDN_SPKLO (1 << 4) +#define CS42L73_PDN_EAR (1 << 3) +#define CS42L73_PDN_SPK (1 << 2) +#define CS42L73_PDN_LO (1 << 1) +#define CS42L73_PDN_HP (1 << 0) /* Thermal Overload Detect. Requires interrupt ... */ -#define THMOVLD_150C 0 -#define THMOVLD_132C 1 -#define THMOVLD_115C 2 -#define THMOVLD_098C 3 +#define CS42L73_THMOVLD_150C 0 +#define CS42L73_THMOVLD_132C 1 +#define CS42L73_THMOVLD_115C 2 +#define CS42L73_THMOVLD_098C 3 #define CS42L73_CHARGEPUMP_MASK (0xF0) /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ -#define SP_3ST (1 << 7) -#define SPDIF_I2S (0 << 6) -#define SPDIF_PCM (1 << 6) -#define PCM_MODE0 (0 << 4) -#define PCM_MODE1 (1 << 4) -#define PCM_MODE2 (2 << 4) -#define PCM_MODE_MASK (3 << 4) -#define PCM_BIT_ORDER (1 << 3) -#define MCK_SCLK_64FS (0 << 0) -#define MCK_SCLK_MCLK (2 << 0) -#define MCK_SCLK_PREMCLK (3 << 0) +#define CS42L73_SP_3ST (1 << 7) +#define CS42L73_SPDIF_I2S (0 << 6) +#define CS42L73_SPDIF_PCM (1 << 6) +#define CS42L73_PCM_MODE0 (0 << 4) +#define CS42L73_PCM_MODE1 (1 << 4) +#define CS42L73_PCM_MODE2 (2 << 4) +#define CS42L73_PCM_MODE_MASK (3 << 4) +#define CS42L73_PCM_BIT_ORDER (1 << 3) +#define CS42L73_MCK_SCLK_64FS (0 << 0) +#define CS42L73_MCK_SCLK_MCLK (2 << 0) +#define CS42L73_MCK_SCLK_PREMCLK (3 << 0) /* CS42L73_xSPMMCC */ -#define MS_MASTER (1 << 7) +#define CS42L73_MS_MASTER (1 << 7) /* CS42L73_DMMCC */ -#define MCLKDIS (1 << 0) -#define MCLKSEL_MCLK2 (1 << 4) -#define MCLKSEL_MCLK1 (0 << 4) +#define CS42L73_MCLKDIS (1 << 0) +#define CS42L73_MCLKSEL_MCLK2 (1 << 4) +#define CS42L73_MCLKSEL_MCLK1 (0 << 4) /* CS42L73 MCLK derived from MCLK1 or MCLK2 */ #define CS42L73_CLKID_MCLK1 0 @@ -195,28 +195,26 @@ #define CS42L73_VSP 2 /* IS1, IM1 */ -#define MIC2_SDET (1 << 6) -#define THMOVLD (1 << 4) -#define DIGMIXOVFL (1 << 3) -#define IPBOVFL (1 << 1) -#define IPAOVFL (1 << 0) +#define CS42L73_MIC2_SDET (1 << 6) +#define CS42L73_THMOVLD (1 << 4) +#define CS42L73_DIGMIXOVFL (1 << 3) +#define CS42L73_IPBOVFL (1 << 1) +#define CS42L73_IPAOVFL (1 << 0) /* Analog Softramp */ -#define ANLGOSFT (1 << 0) +#define CS42L73_ANLGOSFT (1 << 0) /* HP A/B Analog Mute */ -#define HPA_MUTE (1 << 7) +#define CS42L73_HPA_MUTE (1 << 7) /* LO A/B Analog Mute */ -#define LOA_MUTE (1 << 7) +#define CS42L73_LOA_MUTE (1 << 7) /* Digital Mute */ -#define HLAD_MUTE (1 << 0) -#define HLBD_MUTE (1 << 1) -#define SPKD_MUTE (1 << 2) -#define ESLD_MUTE (1 << 3) +#define CS42L73_HLAD_MUTE (1 << 0) +#define CS42L73_HLBD_MUTE (1 << 1) +#define CS42L73_SPKD_MUTE (1 << 2) +#define CS42L73_ESLD_MUTE (1 << 3) /* Misc defines for codec */ -#define CS42L73_RESET_GPIO 143 - #define CS42L73_DEVID 0x00042A73 #define CS42L73_MCLKX_MIN 5644800 #define CS42L73_MCLKX_MAX 38400000 -- cgit From 6833c452c2fb47353566aa705d68541c6045c796 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 16 Oct 2013 22:05:26 -0700 Subject: ASoC: add snd_soc_of_get_dai_name() default of_xlate Current snd_soc_of_get_dai_name() needs .of_xlate_dai_name() callback on each component drivers. But required behavior on almost all these drivers is just returns its indexed driver's name. This patch adds this feature as default behavior. .of_xlate_dai_name() can overwrite it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 28 ++++++++++++++++++++++++---- 2 files changed, 27 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index b13eecbaea78..6ed3dc0773cc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -644,10 +644,12 @@ struct snd_soc_component_driver { struct snd_soc_component { const char *name; int id; - int num_dai; struct device *dev; struct list_head list; + struct snd_soc_dai_driver *dai_drv; + int num_dai; + const struct snd_soc_component_driver *driver; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 67cfb5f5ca96..0860a7f11299 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4023,6 +4023,7 @@ __snd_soc_register_component(struct device *dev, cmpnt->dev = dev; cmpnt->driver = cmpnt_drv; + cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; /* @@ -4548,12 +4549,31 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, if (pos->dev->of_node != args.np) continue; - if (!pos->driver->of_xlate_dai_name) { - ret = -ENOSYS; - break; + if (pos->driver->of_xlate_dai_name) { + ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name); + } else { + int id = -1; + + switch (args.args_count) { + case 0: + id = 0; /* same as dai_drv[0] */ + break; + case 1: + id = args.args[0]; + break; + default: + /* not supported */ + break; + } + + if (id < 0 || id >= pos->num_dai) { + ret = -EINVAL; + } else { + *dai_name = pos->dai_drv[id].name; + ret = 0; + } } - ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name); break; } mutex_unlock(&client_mutex); -- cgit From f1b4863a871d1ca2984bdaa8c88598690530a65c Mon Sep 17 00:00:00 2001 From: "Geyslan G. Bem" Date: Thu, 17 Oct 2013 19:57:12 -0300 Subject: ALSA: emu10k1: code refactoring Partially restructures _snd_emu10k1_audigy_init_efx() and _snd_emu10k1_init_efx() functions. Be noted that the cast is demanded to use '__user'. So, in these cases, avoid patches based on the coccinelle 'drop_kmalloc_cast' semantic patch. Signed-off-by: Geyslan G. Bem Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emufx.c | 76 ++++++++++++++++++++++++++++------------------- 1 file changed, 45 insertions(+), 31 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 0275209ca82e..1f9c7c4bbcd8 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1182,15 +1182,20 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL || - (icode->gpr_map = (u_int32_t __user *) - kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(*controls), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (u_int32_t __user *) kcalloc(512 + 256 + 256 + 2 * 1024, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(*controls), GFP_KERNEL); + if (!controls) + goto __err_ctrls; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 512; @@ -1741,12 +1746,12 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) emu->support_tlv = 0; /* clear again */ snd_leave_user(seg); - __err: +__err: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } @@ -1813,18 +1818,26 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL) - return -ENOMEM; - if ((icode->gpr_map = (u_int32_t __user *) - kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(struct snd_emu10k1_fx8010_control_gpr), - GFP_KERNEL)) == NULL || - (ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (u_int32_t __user *) kcalloc(256 + 160 + 160 + 2 * 512, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(struct snd_emu10k1_fx8010_control_gpr), + GFP_KERNEL); + if (!controls) + goto __err_ctrls; + + ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL); + if (!ipcm) + goto __err_ipcm; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 256; @@ -2363,13 +2376,14 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_leave_user(seg); if (err >= 0) err = snd_emu10k1_ipcm_poke(emu, ipcm); - __err: +__err: kfree(ipcm); +__err_ipcm: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } -- cgit From ca50410b731c636b9750c02d5ae45be215056634 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Oct 2013 14:56:13 +0100 Subject: ASoC: wm8962: Move interrupt initalisation to probe() This is more idiomatic and fixes bugs in the error handling paths. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 68 +++++++++++++++++++++++------------------------ 1 file changed, 33 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11d80f3b6137..54379ee1cd0c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = &wm8962->pdata; - int i, trigger, irq_pol; + int i; bool dmicclk, dmicdat; wm8962->codec = codec; @@ -3506,36 +3506,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_init_beep(codec); wm8962_init_gpio(codec); - if (wm8962->irq) { - if (pdata->irq_active_low) { - trigger = IRQF_TRIGGER_LOW; - irq_pol = WM8962_IRQ_POL; - } else { - trigger = IRQF_TRIGGER_HIGH; - irq_pol = 0; - } - - snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL, - WM8962_IRQ_POL, irq_pol); - - ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq, - trigger | IRQF_ONESHOT, - "wm8962", codec->dev); - if (ret != 0) { - dev_err(codec->dev, "Failed to request IRQ %d: %d\n", - wm8962->irq, ret); - wm8962->irq = 0; - /* Non-fatal */ - } else { - /* Enable some IRQs by default */ - snd_soc_update_bits(codec, - WM8962_INTERRUPT_STATUS_2_MASK, - WM8962_FLL_LOCK_EINT | - WM8962_TEMP_SHUT_EINT | - WM8962_FIFOS_ERR_EINT, 0); - } - } - return 0; } @@ -3544,9 +3514,6 @@ static int wm8962_remove(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; - if (wm8962->irq) - free_irq(wm8962->irq, codec); - cancel_delayed_work_sync(&wm8962->mic_work); wm8962_free_gpio(codec); @@ -3619,7 +3586,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, struct wm8962_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm8962_priv *wm8962; unsigned int reg; - int ret, i; + int ret, i, irq_pol, trigger; wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv), GFP_KERNEL); @@ -3714,6 +3681,37 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, ret); } + if (wm8962->irq) { + if (pdata->irq_active_low) { + trigger = IRQF_TRIGGER_LOW; + irq_pol = WM8962_IRQ_POL; + } else { + trigger = IRQF_TRIGGER_HIGH; + irq_pol = 0; + } + + regmap_update_bits(wm8962->regmap, WM8962_INTERRUPT_CONTROL, + WM8962_IRQ_POL, irq_pol); + + ret = devm_request_threaded_irq(&i2c->dev, wm8962->irq, NULL, + wm8962_irq, + trigger | IRQF_ONESHOT, + "wm8962", &i2c->dev); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request IRQ %d: %d\n", + wm8962->irq, ret); + wm8962->irq = 0; + /* Non-fatal */ + } else { + /* Enable some IRQs by default */ + regmap_update_bits(wm8962->regmap, + WM8962_INTERRUPT_STATUS_2_MASK, + WM8962_FLL_LOCK_EINT | + WM8962_TEMP_SHUT_EINT | + WM8962_FIFOS_ERR_EINT, 0); + } + } + pm_runtime_enable(&i2c->dev); pm_request_idle(&i2c->dev); -- cgit From 78b78f5c019e5c68c88afad4b0d3070becde939e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Oct 2013 15:04:21 +0100 Subject: ASoC: wm8962: Move register initialisation to I2C probe() This is more idiomatic and is required for robust operation since we must ensure that the clocking configuration is valid as rapidly as possible. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 150 +++++++++++++++++++++++----------------------- 1 file changed, 75 insertions(+), 75 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 54379ee1cd0c..2bf9ee7c5407 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3242,7 +3242,7 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) } #endif -static void wm8962_set_gpio_mode(struct snd_soc_codec *codec, int gpio) +static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) { int mask = 0; int val = 0; @@ -3263,8 +3263,8 @@ static void wm8962_set_gpio_mode(struct snd_soc_codec *codec, int gpio) } if (mask) - snd_soc_update_bits(codec, WM8962_ANALOGUE_CLOCKING1, - mask, val); + regmap_update_bits(wm8962->regmap, WM8962_ANALOGUE_CLOCKING1, + mask, val); } #ifdef CONFIG_GPIOLIB @@ -3276,7 +3276,6 @@ static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip) static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) { struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); - struct snd_soc_codec *codec = wm8962->codec; /* The WM8962 GPIOs aren't linearly numbered. For simplicity * we export linear numbers and error out if the unsupported @@ -3292,7 +3291,7 @@ static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) return -EINVAL; } - wm8962_set_gpio_mode(codec, offset + 1); + wm8962_set_gpio_mode(wm8962, offset + 1); return 0; } @@ -3376,7 +3375,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = &wm8962->pdata; int i; bool dmicclk, dmicdat; @@ -3409,75 +3407,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) } } - /* SYSCLK defaults to on; make sure it is off so we can safely - * write to registers if the device is declocked. - */ - snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); - - /* Ensure we have soft control over all registers */ - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); - - /* Ensure that the oscillator and PLLs are disabled */ - snd_soc_update_bits(codec, WM8962_PLL2, - WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, - 0); - - /* Apply static configuration for GPIOs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) - if (pdata->gpio_init[i]) { - wm8962_set_gpio_mode(codec, i + 1); - snd_soc_write(codec, 0x200 + i, - pdata->gpio_init[i] & 0xffff); - } - - - /* Put the speakers into mono mode? */ - if (pdata->spk_mono) - snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, - WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); - - /* Micbias setup, detection enable and detection - * threasholds. */ - if (pdata->mic_cfg) - snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4, - WM8962_MICDET_ENA | - WM8962_MICDET_THR_MASK | - WM8962_MICSHORT_THR_MASK | - WM8962_MICBIAS_LVL, - pdata->mic_cfg); - - /* Latch volume update bits */ - snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, - WM8962_IN_VU, WM8962_IN_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, - WM8962_IN_VU, WM8962_IN_VU); - snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, - WM8962_ADC_VU, WM8962_ADC_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, - WM8962_ADC_VU, WM8962_ADC_VU); - snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, - WM8962_DAC_VU, WM8962_DAC_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, - WM8962_DAC_VU, WM8962_DAC_VU); - snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, - WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); - snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, - WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); - snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, - WM8962_HPOUT_VU, WM8962_HPOUT_VU); - snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, - WM8962_HPOUT_VU, WM8962_HPOUT_VU); - - /* Stereo control for EQ */ - snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0); - - /* Don't debouce interrupts so we don't need SYSCLK */ - snd_soc_update_bits(codec, WM8962_IRQ_DEBOUNCE, - WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | - WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, - 0); - wm8962_add_widgets(codec); /* Save boards having to disable DMIC when not in use */ @@ -3671,6 +3600,77 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, goto err_enable; } + /* SYSCLK defaults to on; make sure it is off so we can safely + * write to registers if the device is declocked. + */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA, 0); + + /* Ensure we have soft control over all registers */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + + /* Ensure that the oscillator and PLLs are disabled */ + regmap_update_bits(wm8962->regmap, WM8962_PLL2, + WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, + 0); + + /* Apply static configuration for GPIOs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) + if (pdata->gpio_init[i]) { + wm8962_set_gpio_mode(wm8962, i + 1); + regmap_write(wm8962->regmap, 0x200 + i, + pdata->gpio_init[i] & 0xffff); + } + + + /* Put the speakers into mono mode? */ + if (pdata->spk_mono) + regmap_update_bits(wm8962->regmap, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); + + /* Micbias setup, detection enable and detection + * threasholds. */ + if (pdata->mic_cfg) + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_MICDET_ENA | + WM8962_MICDET_THR_MASK | + WM8962_MICSHORT_THR_MASK | + WM8962_MICBIAS_LVL, + pdata->mic_cfg); + + /* Latch volume update bits */ + regmap_update_bits(wm8962->regmap, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + regmap_update_bits(wm8962->regmap, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + regmap_update_bits(wm8962->regmap, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + regmap_update_bits(wm8962->regmap, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + + /* Stereo control for EQ */ + regmap_update_bits(wm8962->regmap, WM8962_EQ1, + WM8962_EQ_SHARED_COEFF, 0); + + /* Don't debouce interrupts so we don't need SYSCLK */ + regmap_update_bits(wm8962->regmap, WM8962_IRQ_DEBOUNCE, + WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | + WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, + 0); + if (wm8962->pdata.in4_dc_measure) { ret = regmap_register_patch(wm8962->regmap, wm8962_dc_measure, -- cgit From 6197c34425c3d9a622d3a7031c91104909224a67 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 14:09:27 +0100 Subject: ASoC: cirrus: Enable compile test builds The core support for ep93xx (currently only the DMA driver) does not depend on the architecture at all and everything else has more strict dependencies so enable compile test builds for improved build coverage. Signed-off-by: Mark Brown --- sound/soc/cirrus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2c20f01e1f7e..06f938deda15 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on ARCH_EP93XX && SND_SOC + depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- cgit From e58f301ec969430cdafd7fa872660458f4939507 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Oct 2013 17:26:22 +0100 Subject: ASoC: rt5640: Power down LDO while suspended If we have control over the LDO then disable it during suspend; the device is already being put into reset so will be non-functional over suspend anyway and this will save a small amount of power. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/rt5640.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 641eeeb00c5c..b0cde92be7eb 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1979,12 +1979,20 @@ static int rt5640_suspend(struct snd_soc_codec *codec) rt5640_reset(codec); regcache_cache_only(rt5640->regmap, true); regcache_mark_dirty(rt5640->regmap); + if (gpio_is_valid(rt5640->pdata.ldo1_en)) + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 0); return 0; } static int rt5640_resume(struct snd_soc_codec *codec) { + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 1); + msleep(400); + } rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- cgit From ebce31140c2ddfda005e88957ac1ee1eacaa8dc5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Oct 2013 17:30:44 +0100 Subject: ASoC: rt5640: Don't go to standby on resume There is no need for the CODEC to go to standby on resume since the core will power it up as needed and in any case it is an idle_bias_off CODEC so would normally sit with bias off while idle. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/rt5640.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b0cde92be7eb..4d041d376f31 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1993,7 +1993,6 @@ static int rt5640_resume(struct snd_soc_codec *codec) gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 1); msleep(400); } - rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } -- cgit From c0de42bf595238e9dd593405ebc2992cc8470732 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 8 Oct 2013 15:07:59 +0200 Subject: ASoC: dmaengine-pcm: Add support for querying DMA capabilities Currently each platform making use the the generic dmaengine PCM driver still needs to provide a custom snd_pcm_hardware struct which specifies the capabilities of the DMA controller, e.g. the maximum period size that can be supported. This patch adds code which uses the newly introduced dma_get_slave_caps() API to query this information from the dmaengine driver. The new code path will only be taken if the 'pcm_hardware' field of the snd_dmaengine_pcm_config struct is NULL. The patch also introduces a new 'fifo_size' field to the snd_dmaengine_dai_dma_data struct which is used to initialize the snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 2 ++ sound/soc/soc-generic-dmaengine-pcm.c | 55 ++++++++++++++++++++++++++++------- 2 files changed, 47 insertions(+), 10 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f11c35cd5532..83b2c3e95a1a 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. + * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -68,6 +69,7 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + unsigned int fifo_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3cd84b1..c39e19e84c8a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -36,6 +36,15 @@ static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) return container_of(p, struct dmaengine_pcm, platform); } +static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, + struct snd_pcm_substream *substream) +{ + if (!pcm->chan[substream->stream]) + return NULL; + + return pcm->chan[substream->stream]->device->dev; +} + /** * snd_dmaengine_pcm_prepare_slave_config() - Generic prepare_slave_config callback * @substream: PCM substream @@ -92,28 +101,54 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } -static int dmaengine_pcm_open(struct snd_pcm_substream *substream) +static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct device *dma_dev = dmaengine_dma_dev(pcm, substream); struct dma_chan *chan = pcm->chan[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data; + struct dma_slave_caps dma_caps; + struct snd_pcm_hardware hw; int ret; - ret = snd_soc_set_runtime_hwparams(substream, + if (pcm->config->pcm_hardware) + return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); - if (ret) - return ret; - return snd_dmaengine_pcm_open(substream, chan); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = UINT_MAX; + hw.period_bytes_min = 256; + hw.period_bytes_max = dma_get_max_seg_size(dma_dev); + hw.buffer_bytes_max = SIZE_MAX; + hw.fifo_size = dma_data->fifo_size; + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret == 0) { + if (dma_caps.cmd_pause) + hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + } + + return snd_soc_set_runtime_hwparams(substream, &hw); } -static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, - struct snd_pcm_substream *substream) +static int dmaengine_pcm_open(struct snd_pcm_substream *substream) { - if (!pcm->chan[substream->stream]) - return NULL; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = pcm->chan[substream->stream]; + int ret; - return pcm->chan[substream->stream]->device->dev; + ret = dmaengine_pcm_set_runtime_hwparams(substream); + if (ret) + return ret; + + return snd_dmaengine_pcm_open(substream, chan); } static void dmaengine_pcm_free(struct snd_pcm *pcm) -- cgit From fa654e085300e9c222ef931bc0702a9df2542666 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 8 Oct 2013 15:08:00 +0200 Subject: ASoC: dmaengine-pcm: Provide default config This patch adds some default settings for the generic dmaengine PCM driver for the case that no config has been supplied. The following defaults are used: * Use snd_dmaengine_pcm_prepare_slave_config for preparing the DMA slave config. * 512kB for the prealloc buffer size. This value has been chosen based on 'feels about right' and is not backed up by any scientific facts. We may need to come up with something smarter in the future but it should work fine for now. With this infrastructure in place we can finally write DAI drivers which are independent of the DMA controller they are connected to. This is e.g. useful if the DAI IP core is reused across different SoCs, but the SoCs uses different DMA controllers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index c39e19e84c8a..99f9495c1c40 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -84,12 +84,19 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + int (*prepare_slave_config)(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config); struct dma_slave_config slave_config; int ret; - if (pcm->config->prepare_slave_config) { - ret = pcm->config->prepare_slave_config(substream, params, - &slave_config); + if (!pcm->config) + prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; + else + prepare_slave_config = pcm->config->prepare_slave_config; + + if (prepare_slave_config) { + ret = prepare_slave_config(substream, params, &slave_config); if (ret) return ret; @@ -112,7 +119,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea struct snd_pcm_hardware hw; int ret; - if (pcm->config->pcm_hardware) + if (pcm->config && pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); @@ -177,9 +184,20 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; struct snd_pcm_substream *substream; + size_t prealloc_buffer_size; + size_t max_buffer_size; unsigned int i; int ret; + if (config && config->prealloc_buffer_size) { + prealloc_buffer_size = config->prealloc_buffer_size; + max_buffer_size = config->pcm_hardware->buffer_bytes_max; + } else { + prealloc_buffer_size = 512 * 1024; + max_buffer_size = SIZE_MAX; + } + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { substream = rtd->pcm->streams[i].substream; if (!substream) @@ -200,8 +218,8 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) ret = snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, dmaengine_dma_dev(pcm, substream), - config->prealloc_buffer_size, - config->pcm_hardware->buffer_bytes_max); + prealloc_buffer_size, + max_buffer_size); if (ret) goto err_free; } -- cgit From ecfc0c04f236f1e2a95094792ec10cf27be39f7c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Oct 2013 21:13:19 +0100 Subject: ASoC: dai: Provide interface for setting DMA data at probe time Allow DMA data to be set at probe time for devices that can do that, avoiding the need to do it every time we start a stream and supporting non-DT dmaengine users using the helpers. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227d35d3..97943fd397cc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -276,6 +276,13 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, dai->capture_dma_data = data; } +static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, + void *playback, void *capture) +{ + dai->playback_dma_data = playback; + dai->capture_dma_data = capture; +} + static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, void *data) { -- cgit From 511e30331745e0c3452b89354a4b94c0e60f15a4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Oct 2013 21:18:40 +0100 Subject: ASoC: samsung: Initialise DMA data at device probe time This is a minor simplification and will help with converting the platform to use the dmaengine helpers. Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b302f3b7a587..3e08b6c0f7ba 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -702,13 +702,6 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, } writel(mod, i2s->addr + I2SMOD); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_set_dma_data(dai, substream, - (void *)&i2s->dma_playback); - else - snd_soc_dai_set_dma_data(dai, substream, - (void *)&i2s->dma_capture); - i2s->frmclk = params_rate(params); return 0; @@ -970,6 +963,8 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); + snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + if (other) { other->addr = i2s->addr; other->clk = i2s->clk; -- cgit From e244bb9bc1883547d44642c99f483c2e57e2a940 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 17 Oct 2013 22:46:49 -0700 Subject: ASoC: simple-card: un-implemented set_fmt is not error Current simple-card returns error if DAI doesn't support .set_fmt callback. But the error is -ENOTSUPP (= not supported), and it is not error. This patch avoids such case Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8c49147db84c..b2fbb7075a6c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -27,6 +27,11 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, if (!ret && daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); + if (ret == -ENOTSUPP) { + dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n"); + ret = 0; + } + if (!ret && set->sysclk) ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); -- cgit From cdcfcac968a1ec648434892b6addd80e66a5a892 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 17 Oct 2013 22:50:59 -0700 Subject: ASoC: rcar: add rsnd_scu_hpbif_is_enable() Current SSI needs RSND_SSI_DEPENDENT flag to decide dependent/independent mode. And SCU needs RSND_SCU_USE_HPBIF flag to decide HPBIF is enable/disable. But these 2 means same things. This patch adds new rsnd_scu_hpbif_is_enable() function, and merges above methods. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 1 - sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/scu.c | 12 +++++++++--- sound/soc/sh/rcar/ssi.c | 8 +++++--- 4 files changed, 15 insertions(+), 7 deletions(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index fe66533e9b7a..6b4211c256b3 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -36,7 +36,6 @@ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) #define RSND_SSI_CLK_FROM_ADG (1 << 30) /* clock parent is master */ #define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */ -#define RSND_SSI_DEPENDENT (1 << 28) /* SSI needs SRU/SCU */ #define RSND_SSI_PLAY (1 << 24) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3868aaf41cc4..5feb67ca2d24 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -281,6 +281,7 @@ int rsnd_scu_probe(struct platform_device *pdev, void rsnd_scu_remove(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id); +bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod); #define rsnd_scu_nr(priv) ((priv)->scu_nr) /* diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 2df2e9150b89..1ab1bce6be7f 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -146,20 +146,26 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, return 0; } +bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod) +{ + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + u32 flags = rsnd_scu_mode_flags(scu); + + return !!(flags & RSND_SCU_USE_HPBIF); +} + static int rsnd_scu_start(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_scu *scu = rsnd_mod_to_scu(mod); struct device *dev = rsnd_priv_to_dev(priv); - u32 flags = rsnd_scu_mode_flags(scu); int ret; /* * SCU will be used if it has RSND_SCU_USE_HPBIF flags */ - if (!(flags & RSND_SCU_USE_HPBIF)) { + if (!rsnd_scu_hpbif_is_enable(mod)) { /* it use PIO transter */ dev_dbg(dev, "%s%d is not used\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fae26d3f79d2..7613256c9840 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -106,6 +106,7 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssi *ssi; + struct rsnd_mod *scu; u32 flags; u32 val; int i; @@ -116,13 +117,14 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, ssiu->ssi_mode0 = 0; for_each_rsnd_ssi(ssi, priv, i) { flags = rsnd_ssi_mode_flags(ssi); + scu = rsnd_scu_mod_get(priv, rsnd_mod_id(&ssi->mod)); /* see also BUSIF_MODE */ - if (!(flags & RSND_SSI_DEPENDENT)) { + if (rsnd_scu_hpbif_is_enable(scu)) { + dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i); + } else { ssiu->ssi_mode0 |= (1 << i); dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i); - } else { - dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i); } } -- cgit From 92eba04e4bcd469518cc759ac1bf1a49acaa5cc1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 17 Oct 2013 22:51:40 -0700 Subject: ASoC: rcar: remove RSND_SSI_CLK_FROM_ADG R-Car sound has clock pin for each SSI, and sometimes, these pins are shared with paired SSI. It may sometimes become "SSI-A clock pin is master" and "SSI-B clock pin is slave", but "SSI-A/B clock pins are shared". SSI-B needs SSI-A clock in this case. Current R-Car sound driver is using RSND_SSI_xxx flag to control this kind of shared pin behavior. But, this information, especially clock master setting, can be got from ASoC set_fmt settings. This patch removes rsnd_ssi_mode_init() and extend rsnd_ssi_mode_set() to controlling pin settings via .set_fmt. This patch doesn't removes RSND_SSI_CLK_FROM_ADG flag at this point to avoid conflict branch merging between ASoC <-> SH-ARM. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 52 +++++++++++++++++++------------------------------ 1 file changed, 20 insertions(+), 32 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 7613256c9840..b71cf9d7dd3f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -101,31 +101,30 @@ struct rsnd_ssiu { #define rsnd_ssi_to_ssiu(ssi)\ (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1) -static void rsnd_ssi_mode_init(struct rsnd_priv *priv, - struct rsnd_ssiu *ssiu) +static void rsnd_ssi_mode_set(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_ssi *ssi) { struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_ssi *ssi; struct rsnd_mod *scu; + struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi); + int id = rsnd_mod_id(&ssi->mod); u32 flags; u32 val; - int i; + + scu = rsnd_scu_mod_get(priv, rsnd_mod_id(&ssi->mod)); /* * SSI_MODE0 */ - ssiu->ssi_mode0 = 0; - for_each_rsnd_ssi(ssi, priv, i) { - flags = rsnd_ssi_mode_flags(ssi); - scu = rsnd_scu_mod_get(priv, rsnd_mod_id(&ssi->mod)); - - /* see also BUSIF_MODE */ - if (rsnd_scu_hpbif_is_enable(scu)) { - dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i); - } else { - ssiu->ssi_mode0 |= (1 << i); - dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i); - } + + /* see also BUSIF_MODE */ + if (rsnd_scu_hpbif_is_enable(scu)) { + ssiu->ssi_mode0 &= ~(1 << id); + dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", id); + } else { + ssiu->ssi_mode0 |= (1 << id); + dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", id); } /* @@ -134,7 +133,7 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, #define ssi_parent_set(p, sync, adg, ext) \ do { \ ssi->parent = ssiu->ssi + p; \ - if (flags & RSND_SSI_CLK_FROM_ADG) \ + if (rsnd_rdai_is_clk_master(rdai)) \ val = adg; \ else \ val = ext; \ @@ -142,15 +141,11 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, val |= sync; \ } while (0) - ssiu->ssi_mode1 = 0; - for_each_rsnd_ssi(ssi, priv, i) { - flags = rsnd_ssi_mode_flags(ssi); - - if (!(flags & RSND_SSI_CLK_PIN_SHARE)) - continue; + flags = rsnd_ssi_mode_flags(ssi); + if (flags & RSND_SSI_CLK_PIN_SHARE) { val = 0; - switch (i) { + switch (id) { case 1: ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0)); break; @@ -167,11 +162,6 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, ssiu->ssi_mode1 |= val; } -} - -static void rsnd_ssi_mode_set(struct rsnd_ssi *ssi) -{ - struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi); rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0); rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1); @@ -381,7 +371,7 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->cr_own = cr; ssi->err = -1; /* ignore 1st error */ - rsnd_ssi_mode_set(ssi); + rsnd_ssi_mode_set(priv, rdai, ssi); dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); @@ -708,8 +698,6 @@ int rsnd_ssi_probe(struct platform_device *pdev, rsnd_mod_init(priv, &ssi->mod, ops, i); } - rsnd_ssi_mode_init(priv, ssiu); - dev_dbg(dev, "ssi probed\n"); return 0; -- cgit From 1f1b65796ef882bb9101d22b17e1a1824b3a6489 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Fri, 18 Oct 2013 20:34:52 +0200 Subject: ASoC: kirkwood: prefer external clock over internal clock When there is an external clock, always use this one. This prevents the two Dove audio devices to use the same DCO clock at different rates. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3e59af983527..d0504a2d8c63 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -103,7 +103,7 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, { uint32_t clks_ctrl; - if (rate == 44100 || rate == 48000 || rate == 96000) { + if (IS_ERR(priv->extclk)) { /* use internal dco for the supported rates * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", -- cgit From 7b09eea52939d2b979f19de40e34b8670feff4c5 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 18 Oct 2013 14:30:01 -0500 Subject: ASoC: cs42l73: Add Device Tree support for CS42L73 This patch adds support for device tree for the CS42L73 CODEC Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/cs42l73.txt | 22 ++++++++++++++++ sound/soc/codecs/cs42l73.c | 29 ++++++++++++++++++++-- 2 files changed, 49 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l73.txt diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt new file mode 100644 index 000000000000..80ae910dbf6c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l73.txt @@ -0,0 +1,22 @@ +CS42L73 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l73" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset_gpio : a GPIO spec for the reset pin. + - chgfreq : Charge Pump Frequency values 0x00-0x0F + + +Example: + +codec: cs42l73@4a { + compatible = "cirrus,cs42l73"; + reg = <0x4a>; + reset_gpio = <&gpio 10 0>; + chgfreq = <0x05>; +}; \ No newline at end of file diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 89efc3c6aefc..549d5d6a3fef 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,7 +17,7 @@ #include #include #include -#include +#include #include #include #include @@ -1416,6 +1416,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, int ret; unsigned int devid = 0; unsigned int reg; + u32 val32; cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), GFP_KERNEL); @@ -1431,8 +1432,25 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return ret; } - if (pdata) + if (pdata) { cs42l73->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l73_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_u32(i2c_client->dev.of_node, + "chgfreq", &val32) >= 0) + pdata->chgfreq = val32; + } + pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, + "reset-gpio", 0); + cs42l73->pdata = *pdata; + } i2c_set_clientdata(i2c_client, cs42l73); @@ -1493,6 +1511,12 @@ static int cs42l73_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l73_of_match[] = { + { .compatible = "cirrus,cs42l73", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l73_of_match); + static const struct i2c_device_id cs42l73_id[] = { {"cs42l73", 0}, {} @@ -1504,6 +1528,7 @@ static struct i2c_driver cs42l73_i2c_driver = { .driver = { .name = "cs42l73", .owner = THIS_MODULE, + .of_match_table = cs42l73_of_match, }, .id_table = cs42l73_id, .probe = cs42l73_i2c_probe, -- cgit From 6abce9e63d44e94b16f08794fb6f5ad6d1025c79 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:11:14 +0200 Subject: ALSA: dice: optimize bus reset handling After a bus reset, do not stop the stream completely to avoid having to reconfigure the device when restarting the stream. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 148 +++++++++++++++++++++++++++++++------------------- 1 file changed, 93 insertions(+), 55 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index ac71b2b94eec..b081021064ed 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -559,24 +559,92 @@ static int dice_close(struct snd_pcm_substream *substream) return 0; } -static void dice_stop_stream(struct dice *dice) +static int dice_stream_start_packets(struct dice *dice) { - __be32 channel; + int err; + + if (dice->stream_running) + return 0; - if (dice->stream_running) { - dice_enable_clear(dice); + err = amdtp_out_stream_start(&dice->stream, dice->resources.channel, + fw_parent_device(dice->unit)->max_speed); + if (err < 0) + return err; + err = dice_enable_set(dice); + if (err < 0) { amdtp_out_stream_stop(&dice->stream); + return err; + } - channel = cpu_to_be32((u32)-1); - snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), - &channel, 4); + dice->stream_running = true; - fw_iso_resources_free(&dice->resources); + return 0; +} - dice->stream_running = false; +static int dice_stream_start(struct dice *dice) +{ + __be32 channel; + int err; + + if (!dice->resources.allocated) { + err = fw_iso_resources_allocate(&dice->resources, + amdtp_out_stream_get_max_payload(&dice->stream), + fw_parent_device(dice->unit)->max_speed); + if (err < 0) + goto error; + + channel = cpu_to_be32(dice->resources.channel); + err = snd_fw_transaction(dice->unit, + TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), + &channel, 4); + if (err < 0) + goto err_resources; } + + err = dice_stream_start_packets(dice); + if (err < 0) + goto err_rx_channel; + + return 0; + +err_rx_channel: + channel = cpu_to_be32((u32)-1); + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), &channel, 4); +err_resources: + fw_iso_resources_free(&dice->resources); +error: + return err; +} + +static void dice_stream_stop_packets(struct dice *dice) +{ + if (!dice->stream_running) + return; + + dice_enable_clear(dice); + + amdtp_out_stream_stop(&dice->stream); + + dice->stream_running = false; +} + +static void dice_stream_stop(struct dice *dice) +{ + __be32 channel; + + dice_stream_stop_packets(dice); + + if (!dice->resources.allocated) + return; + + channel = cpu_to_be32((u32)-1); + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), &channel, 4); + + fw_iso_resources_free(&dice->resources); } static int dice_hw_params(struct snd_pcm_substream *substream, @@ -586,7 +654,7 @@ static int dice_hw_params(struct snd_pcm_substream *substream, int err; mutex_lock(&dice->mutex); - dice_stop_stream(dice); + dice_stream_stop(dice); mutex_unlock(&dice->mutex); err = snd_pcm_lib_alloc_vmalloc_buffer(substream, @@ -608,7 +676,7 @@ static int dice_hw_free(struct snd_pcm_substream *substream) struct dice *dice = substream->private_data; mutex_lock(&dice->mutex); - dice_stop_stream(dice); + dice_stream_stop(dice); mutex_unlock(&dice->mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -617,42 +685,17 @@ static int dice_hw_free(struct snd_pcm_substream *substream) static int dice_prepare(struct snd_pcm_substream *substream) { struct dice *dice = substream->private_data; - struct fw_device *device = fw_parent_device(dice->unit); - __be32 channel; int err; mutex_lock(&dice->mutex); if (amdtp_out_streaming_error(&dice->stream)) - dice_stop_stream(dice); - - if (!dice->stream_running) { - err = fw_iso_resources_allocate(&dice->resources, - amdtp_out_stream_get_max_payload(&dice->stream), - device->max_speed); - if (err < 0) - goto error; - - //TODO: RX_SEQ_START - channel = cpu_to_be32(dice->resources.channel); - err = snd_fw_transaction(dice->unit, - TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), - &channel, 4); - if (err < 0) - goto err_resources; - - err = amdtp_out_stream_start(&dice->stream, - dice->resources.channel, - device->max_speed); - if (err < 0) - goto err_resources; - - err = dice_enable_set(dice); - if (err < 0) - goto err_stream; + dice_stream_stop_packets(dice); - dice->stream_running = true; + err = dice_stream_start(dice); + if (err < 0) { + mutex_unlock(&dice->mutex); + return err; } mutex_unlock(&dice->mutex); @@ -660,15 +703,6 @@ static int dice_prepare(struct snd_pcm_substream *substream) amdtp_out_stream_pcm_prepare(&dice->stream); return 0; - -err_stream: - amdtp_out_stream_stop(&dice->stream); -err_resources: - fw_iso_resources_free(&dice->resources); -error: - mutex_unlock(&dice->mutex); - - return err; } static int dice_trigger(struct snd_pcm_substream *substream, int cmd) @@ -941,7 +975,7 @@ static void dice_remove(struct fw_unit *unit) mutex_lock(&dice->mutex); amdtp_out_stream_pcm_abort(&dice->stream); - dice_stop_stream(dice); + dice_stream_stop(dice); dice_owner_clear(dice); mutex_unlock(&dice->mutex); @@ -953,8 +987,8 @@ static void dice_bus_reset(struct fw_unit *unit) struct dice *dice = dev_get_drvdata(&unit->device); mutex_lock(&dice->mutex); + /* - * XXX is the following true? * On a bus reset, the DICE firmware disables streaming and then goes * off contemplating its own navel for hundreds of milliseconds before * it can react to any of our attempts to reenable streaming. This @@ -962,9 +996,13 @@ static void dice_bus_reset(struct fw_unit *unit) * to stop so that the application can restart them in an orderly * manner. */ - dice_owner_update(dice); amdtp_out_stream_pcm_abort(&dice->stream); - dice_stop_stream(dice); + dice_stream_stop_packets(dice); + + dice_owner_update(dice); + + fw_iso_resources_update(&dice->resources); + mutex_unlock(&dice->mutex); } -- cgit From 341682cd4f5603fccbf559d201402539880c04a5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:12:06 +0200 Subject: ALSA: dice: allow all sample rates Instead of forcing a constant 44.1 kHz, read the current sample rate from the device when opening the PCM device. Actually changing the sample rate requires some separate controller application. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 55 +++++++++++++++++++++++++++++++++++++++------------ 1 file changed, 42 insertions(+), 13 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index b081021064ed..d3f3eb7a546a 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -75,6 +75,7 @@ #define CLOCK_RATE_ANY_MID 0x00000800 #define CLOCK_RATE_ANY_HIGH 0x00000900 #define CLOCK_RATE_NONE 0x00000a00 +#define CLOCK_RATE_SHIFT 8 #define GLOBAL_ENABLE 0x050 #define ENABLE 0x00000001 #define GLOBAL_STATUS 0x054 @@ -248,6 +249,16 @@ MODULE_DESCRIPTION("DICE driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); +static const unsigned int dice_rates[] = { + [0] = 32000, + [1] = 44100, + [2] = 48000, + [3] = 88200, + [4] = 96000, + [5] = 176400, + [6] = 192000, +}; + static inline u64 global_address(struct dice *dice, unsigned int offset) { return DICE_PRIVATE_SPACE + dice->global_offset + offset; @@ -508,9 +519,6 @@ static int dice_open(struct snd_pcm_substream *substream) SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .rates = SNDRV_PCM_RATE_44100, - .rate_min = 44100, - .rate_max = 44100, .buffer_bytes_max = 16 * 1024 * 1024, .period_bytes_min = 1, .period_bytes_max = UINT_MAX, @@ -519,9 +527,20 @@ static int dice_open(struct snd_pcm_substream *substream) }; struct dice *dice = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - __be32 number_audio, number_midi; + __be32 clock_sel, number_audio, number_midi; + unsigned int rate; int err; + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &clock_sel, 4); + if (err < 0) + return err; + rate = (be32_to_cpu(clock_sel) & CLOCK_RATE_MASK) >> CLOCK_RATE_SHIFT; + if (rate >= ARRAY_SIZE(dice_rates)) + return -ENXIO; + rate = dice_rates[rate]; + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, rx_address(dice, RX_NUMBER_AUDIO), &number_audio, 4); @@ -534,10 +553,14 @@ static int dice_open(struct snd_pcm_substream *substream) return err; runtime->hw = hardware; + + runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate); + snd_pcm_limit_hw_rates(runtime); + runtime->hw.channels_min = be32_to_cpu(number_audio); runtime->hw.channels_max = be32_to_cpu(number_audio); - amdtp_out_stream_set_rate(&dice->stream, 44100); + amdtp_out_stream_set_rate(&dice->stream, rate); amdtp_out_stream_set_pcm(&dice->stream, be32_to_cpu(number_audio)); amdtp_out_stream_set_midi(&dice->stream, be32_to_cpu(number_midi)); @@ -746,17 +769,9 @@ static int dice_create_pcm(struct dice *dice) .page = snd_pcm_lib_get_vmalloc_page, .mmap = snd_pcm_lib_mmap_vmalloc, }; - __be32 clock; struct snd_pcm *pcm; int err; - clock = cpu_to_be32(CLOCK_SOURCE_ARX1 | CLOCK_RATE_44100); - err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &clock, 4); - if (err < 0) - return err; - err = snd_pcm_new(dice->card, "DICE", 0, 1, 0, &pcm); if (err < 0) return err; @@ -897,6 +912,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { struct snd_card *card; struct dice *dice; + __be32 clock_sel; int err; err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*dice), &card); @@ -938,6 +954,19 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice_card_strings(dice); + err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &clock_sel, 4); + if (err < 0) + goto error; + clock_sel &= cpu_to_be32(~CLOCK_SOURCE_MASK); + clock_sel |= cpu_to_be32(CLOCK_SOURCE_ARX1); + err = snd_fw_transaction(unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &clock_sel, 4); + if (err < 0) + goto error; + err = dice_create_pcm(dice); if (err < 0) goto error; -- cgit From d13109673ac49cd4b992df17238ee030be7ed7f0 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:12:20 +0200 Subject: ALSA: dice: reduce noisy logging The notification bits are not of general interest; log them only when debugging. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index d3f3eb7a546a..02c7b5a42373 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -505,8 +505,8 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, fw_send_response(card, request, RCODE_ADDRESS_ERROR); return; } - dev_info(&dice->unit->device, - "notification: %08x\n", be32_to_cpup(data)); + dev_dbg(&dice->unit->device, + "notification: %08x\n", be32_to_cpup(data)); fw_send_response(card, request, RCODE_COMPLETE); } -- cgit From e84d15f619c13e83b33023c84527ee35ef01b6b4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:12:48 +0200 Subject: ALSA: dice, firewire-lib: add blocking mode Allow AMDTP output streams to use blocking mode. Use it for DICE devices, because the old DICE-II chip will in some cases not be able to lock to non-blocking streams (erratum E7). Signed-off-by: Clemens Ladisch --- sound/firewire/amdtp.c | 54 +++++++++++++++++++++++++++----------------------- sound/firewire/amdtp.h | 7 ++++++- sound/firewire/dice.c | 2 +- 3 files changed, 36 insertions(+), 27 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index ea995af6d049..efb2e2947ce7 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -42,9 +42,6 @@ static void pcm_period_tasklet(unsigned long data); int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit, enum cip_out_flags flags) { - if (flags != CIP_NONBLOCKING) - return -EINVAL; - s->unit = fw_unit_get(unit); s->flags = flags; s->context = ERR_PTR(-1); @@ -96,12 +93,20 @@ void amdtp_out_stream_set_rate(struct amdtp_out_stream *s, unsigned int rate) return; for (sfc = 0; sfc < ARRAY_SIZE(rate_info); ++sfc) - if (rate_info[sfc].rate == rate) { - s->sfc = sfc; - s->syt_interval = rate_info[sfc].syt_interval; - return; - } + if (rate_info[sfc].rate == rate) + goto sfc_found; WARN_ON(1); + return; + +sfc_found: + s->sfc = sfc; + s->syt_interval = rate_info[sfc].syt_interval; + + /* default buffering in the device */ + s->transfer_delay = TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; + if (s->flags & CIP_BLOCKING) + /* additional buffering needed to adjust for no-data packets */ + s->transfer_delay += TICKS_PER_SECOND * s->syt_interval / rate; } EXPORT_SYMBOL(amdtp_out_stream_set_rate); @@ -110,25 +115,15 @@ EXPORT_SYMBOL(amdtp_out_stream_set_rate); * @s: the AMDTP output stream * * This function must not be called before the stream has been configured - * with amdtp_out_stream_set_hw_params(), amdtp_out_stream_set_pcm(), and + * with amdtp_out_stream_set_rate(), amdtp_out_stream_set_pcm(), and * amdtp_out_stream_set_midi(). */ unsigned int amdtp_out_stream_get_max_payload(struct amdtp_out_stream *s) { - static const unsigned int max_data_blocks[] = { - [CIP_SFC_32000] = 4, - [CIP_SFC_44100] = 6, - [CIP_SFC_48000] = 6, - [CIP_SFC_88200] = 12, - [CIP_SFC_96000] = 12, - [CIP_SFC_176400] = 23, - [CIP_SFC_192000] = 24, - }; - s->data_block_quadlets = s->pcm_channels; s->data_block_quadlets += DIV_ROUND_UP(s->midi_ports, 8); - return 8 + max_data_blocks[s->sfc] * 4 * s->data_block_quadlets; + return 8 + s->syt_interval * s->data_block_quadlets * 4; } EXPORT_SYMBOL(amdtp_out_stream_get_max_payload); @@ -248,7 +243,7 @@ static unsigned int calculate_syt(struct amdtp_out_stream *s, s->last_syt_offset = syt_offset; if (syt_offset < TICKS_PER_CYCLE) { - syt_offset += TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; + syt_offset += s->transfer_delay; syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12; syt += syt_offset % TICKS_PER_CYCLE; @@ -344,8 +339,17 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle) return; index = s->packet_index; - data_blocks = calculate_data_blocks(s); syt = calculate_syt(s, cycle); + if (!(s->flags & CIP_BLOCKING)) { + data_blocks = calculate_data_blocks(s); + } else { + if (syt != 0xffff) { + data_blocks = s->syt_interval; + } else { + data_blocks = 0; + syt = 0xffffff; + } + } buffer = s->buffer.packets[index].buffer; buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | @@ -455,9 +459,9 @@ static int queue_initial_skip_packets(struct amdtp_out_stream *s) * @speed: firewire speed code * * The stream cannot be started until it has been configured with - * amdtp_out_stream_set_hw_params(), amdtp_out_stream_set_pcm(), and - * amdtp_out_stream_set_midi(); and it must be started before any - * PCM or MIDI device can be started. + * amdtp_out_stream_set_rate(), amdtp_out_stream_set_pcm(), + * amdtp_out_stream_set_midi(), and amdtp_out_stream_set_format(); + * and it must be started before any PCM or MIDI device can be started. */ int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed) { diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index f6103d68c4b1..fd4ce3030279 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -11,9 +11,13 @@ * sample_rate/8000 samples, with rounding up or down to adjust * for clock skew and left-over fractional samples. This should * be used if supported by the device. + * @CIP_BLOCKING: In blocking mode, each packet contains either zero or + * SYT_INTERVAL samples, with these two types alternating so that + * the overall sample rate comes out right. */ enum cip_out_flags { - CIP_NONBLOCKING = 0, + CIP_NONBLOCKING = 0x00, + CIP_BLOCKING = 0x01, }; /** @@ -51,6 +55,7 @@ struct amdtp_out_stream { __be32 *buffer, unsigned int frames); unsigned int syt_interval; + unsigned int transfer_delay; unsigned int source_node_id_field; struct iso_packets_buffer buffer; diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 02c7b5a42373..63446f86cdd7 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -942,7 +942,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) goto err_notification_handler; dice->resources.channels_mask = 0x00000000ffffffffuLL; - err = amdtp_out_stream_init(&dice->stream, unit, CIP_NONBLOCKING); + err = amdtp_out_stream_init(&dice->stream, unit, CIP_BLOCKING); if (err < 0) goto err_resources; -- cgit From 4ed31f20bb5bb90f003c91734c6b9d18169ae27e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:13:09 +0200 Subject: ALSA: dice: fix hang when unplugging a running device When aborting a PCM stream, the xrun is signaled only if the stream is running. When disconnecting a PCM stream, calling snd_card_disconnect() too early would change the stream into a non-running state and thus prevent the xrun from being noticed by user space. To prevent this, move the snd_card_disconnect() call after the xrun. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 63446f86cdd7..d0575a96ea70 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -1000,12 +1000,15 @@ static void dice_remove(struct fw_unit *unit) { struct dice *dice = dev_get_drvdata(&unit->device); - snd_card_disconnect(dice->card); - mutex_lock(&dice->mutex); + amdtp_out_stream_pcm_abort(&dice->stream); + + snd_card_disconnect(dice->card); + dice_stream_stop(dice); dice_owner_clear(dice); + mutex_unlock(&dice->mutex); snd_card_free_when_closed(dice->card); -- cgit From 0c29c9180fe14b0abb4bfc68b37dda66254689b3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:14:15 +0200 Subject: ALSA: dice: implement hwdep device Implement the hwdep locking and notification mechanisms. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 225 +++++++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 203 insertions(+), 22 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index d0575a96ea70..7225878a09cc 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -5,6 +5,7 @@ * Licensed under the terms of the GNU General Public License, version 2. */ +#include #include #include #include @@ -13,8 +14,11 @@ #include #include #include +#include +#include #include #include +#include #include #include #include @@ -233,13 +237,18 @@ struct dice { struct snd_card *card; struct fw_unit *unit; + spinlock_t lock; struct mutex mutex; unsigned int global_offset; unsigned int rx_offset; struct fw_address_handler notification_handler; int owner_generation; + int dev_lock_count; /* > 0 driver, < 0 userspace */ + bool dev_lock_changed; bool global_enabled; bool stream_running; + wait_queue_head_t hwdep_wait; + u32 notification_bits; struct snd_pcm_substream *pcm; struct fw_iso_resources resources; struct amdtp_out_stream stream; @@ -259,6 +268,47 @@ static const unsigned int dice_rates[] = { [6] = 192000, }; +static void dice_lock_changed(struct dice *dice) +{ + dice->dev_lock_changed = true; + wake_up(&dice->hwdep_wait); +} + +static int dice_try_lock(struct dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count < 0) { + err = -EBUSY; + goto out; + } + + if (dice->dev_lock_count++ == 0) + dice_lock_changed(dice); + err = 0; + +out: + spin_unlock_irq(&dice->lock); + + return err; +} + +static void dice_unlock(struct dice *dice) +{ + spin_lock_irq(&dice->lock); + + if (WARN_ON(dice->dev_lock_count <= 0)) + goto out; + + if (--dice->dev_lock_count == 0) + dice_lock_changed(dice); + +out: + spin_unlock_irq(&dice->lock); +} + static inline u64 global_address(struct dice *dice, unsigned int offset) { return DICE_PRIVATE_SPACE + dice->global_offset + offset; @@ -496,6 +546,7 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, void *data, size_t length, void *callback_data) { struct dice *dice = callback_data; + unsigned long flags; if (tcode != TCODE_WRITE_QUADLET_REQUEST) { fw_send_response(card, request, RCODE_TYPE_ERROR); @@ -505,9 +556,11 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, fw_send_response(card, request, RCODE_ADDRESS_ERROR); return; } - dev_dbg(&dice->unit->device, - "notification: %08x\n", be32_to_cpup(data)); + spin_lock_irqsave(&dice->lock, flags); + dice->notification_bits |= be32_to_cpup(data); + spin_unlock_irqrestore(&dice->lock, flags); fw_send_response(card, request, RCODE_COMPLETE); + wake_up(&dice->hwdep_wait); } static int dice_open(struct snd_pcm_substream *substream) @@ -531,26 +584,32 @@ static int dice_open(struct snd_pcm_substream *substream) unsigned int rate; int err; + err = dice_try_lock(dice); + if (err < 0) + goto error; + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, global_address(dice, GLOBAL_CLOCK_SELECT), &clock_sel, 4); if (err < 0) - return err; + goto err_lock; rate = (be32_to_cpu(clock_sel) & CLOCK_RATE_MASK) >> CLOCK_RATE_SHIFT; - if (rate >= ARRAY_SIZE(dice_rates)) - return -ENXIO; + if (rate >= ARRAY_SIZE(dice_rates)) { + err = -ENXIO; + goto err_lock; + } rate = dice_rates[rate]; err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, rx_address(dice, RX_NUMBER_AUDIO), &number_audio, 4); if (err < 0) - return err; + goto err_lock; err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, rx_address(dice, RX_NUMBER_MIDI), &number_midi, 4); if (err < 0) - return err; + goto err_lock; runtime->hw = hardware; @@ -568,17 +627,26 @@ static int dice_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5000, 8192000); if (err < 0) - return err; + goto err_lock; err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); if (err < 0) - return err; + goto err_lock; return 0; + +err_lock: + dice_unlock(dice); +error: + return err; } static int dice_close(struct snd_pcm_substream *substream) { + struct dice *dice = substream->private_data; + + dice_unlock(dice); + return 0; } @@ -783,45 +851,156 @@ static int dice_create_pcm(struct dice *dice) return 0; } -// TODO: implement these - static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, loff_t *offset) { - return -EIO; + struct dice *dice = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&dice->lock); + + while (!dice->dev_lock_changed && dice->notification_bits == 0) { + prepare_to_wait(&dice->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&dice->lock); + schedule(); + finish_wait(&dice->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&dice->lock); + } + + memset(&event, 0, sizeof(event)); + if (dice->dev_lock_changed) { + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = dice->dev_lock_count > 0; + dice->dev_lock_changed = false; + + count = min(count, (long)sizeof(event.lock_status)); + } else { + event.dice_notification.type = SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION; + event.dice_notification.notification = dice->notification_bits; + dice->notification_bits = 0; + + count = min(count, (long)sizeof(event.dice_notification)); + } + + spin_unlock_irq(&dice->lock); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; } -static int dice_hwdep_open(struct snd_hwdep *hwdep, struct file *file) +static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) { - return -EIO; + struct dice *dice = hwdep->private_data; + unsigned int events; + + poll_wait(file, &dice->hwdep_wait, wait); + + spin_lock_irq(&dice->lock); + if (dice->dev_lock_changed || dice->notification_bits != 0) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&dice->lock); + + return events; } -static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) +static int dice_hwdep_get_info(struct dice *dice, void __user *arg) { + struct fw_device *dev = fw_parent_device(dice->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_DICE; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + return 0; } -static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, - poll_table *wait) +static int dice_hwdep_lock(struct dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count == 0) { + dice->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&dice->lock); + + return err; +} + +static int dice_hwdep_unlock(struct dice *dice) { - return POLLERR | POLLHUP; + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count == -1) { + dice->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&dice->lock); + + return err; } static int dice_hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, unsigned int cmd, unsigned long arg) { - return -EIO; + struct dice *dice = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return dice_hwdep_get_info(dice, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return dice_hwdep_lock(dice); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return dice_hwdep_unlock(dice); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int dice_hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return dice_hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); } +#else +#define dice_hwdep_compat_ioctl NULL +#endif static int dice_create_hwdep(struct dice *dice) { static const struct snd_hwdep_ops ops = { .read = dice_hwdep_read, - .open = dice_hwdep_open, - .release = dice_hwdep_release, .poll = dice_hwdep_poll, .ioctl = dice_hwdep_ioctl, - .ioctl_compat = dice_hwdep_ioctl, + .ioctl_compat = dice_hwdep_compat_ioctl, }; struct snd_hwdep *hwdep; int err; @@ -922,8 +1101,10 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice = card->private_data; dice->card = card; + spin_lock_init(&dice->lock); mutex_init(&dice->mutex); dice->unit = unit; + init_waitqueue_head(&dice->hwdep_wait); err = dice_init_offsets(dice); if (err < 0) -- cgit From 9dd81e3143b57d1bf7e8377ab29b86090baa55a8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:14:54 +0200 Subject: ALSA: dice: clear device lock when closing hwdep device Ensure that misbehaving or aborted userspace programs do not accidentally keep the lock. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 7225878a09cc..ef04089bde7f 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -966,6 +966,18 @@ static int dice_hwdep_unlock(struct dice *dice) return err; } +static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct dice *dice = hwdep->private_data; + + spin_lock_irq(&dice->lock); + if (dice->dev_lock_count == -1) + dice->dev_lock_count = 0; + spin_unlock_irq(&dice->lock); + + return 0; +} + static int dice_hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, unsigned int cmd, unsigned long arg) { @@ -998,6 +1010,7 @@ static int dice_create_hwdep(struct dice *dice) { static const struct snd_hwdep_ops ops = { .read = dice_hwdep_read, + .release = dice_hwdep_release, .poll = dice_hwdep_poll, .ioctl = dice_hwdep_ioctl, .ioctl_compat = dice_hwdep_compat_ioctl, -- cgit From 20b65dd040ce38e2bc0fa3cae13b954c865b61fe Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:15:44 +0200 Subject: ALSA: firewire: introduce amdtp_out_stream_running() Introduce the helper function amdtp_out_stream_running(). This makes many checks in amdtp.c clearer and frees the device drivers from having to track this with a separate variable. Signed-off-by: Clemens Ladisch --- sound/firewire/amdtp.c | 10 +++++----- sound/firewire/amdtp.h | 6 ++++++ sound/firewire/dice.c | 17 +++++------------ sound/firewire/speakers.c | 8 ++------ 4 files changed, 18 insertions(+), 23 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index efb2e2947ce7..d56b8e736b7d 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -59,7 +59,7 @@ EXPORT_SYMBOL(amdtp_out_stream_init); */ void amdtp_out_stream_destroy(struct amdtp_out_stream *s) { - WARN_ON(!IS_ERR(s->context)); + WARN_ON(amdtp_out_stream_running(s)); mutex_destroy(&s->mutex); fw_unit_put(s->unit); } @@ -89,7 +89,7 @@ void amdtp_out_stream_set_rate(struct amdtp_out_stream *s, unsigned int rate) }; unsigned int sfc; - if (WARN_ON(!IS_ERR(s->context))) + if (WARN_ON(amdtp_out_stream_running(s))) return; for (sfc = 0; sfc < ARRAY_SIZE(rate_info); ++sfc) @@ -145,7 +145,7 @@ static void amdtp_write_s32(struct amdtp_out_stream *s, void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s, snd_pcm_format_t format) { - if (WARN_ON(!IS_ERR(s->context))) + if (WARN_ON(amdtp_out_stream_running(s))) return; switch (format) { @@ -481,7 +481,7 @@ int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed) mutex_lock(&s->mutex); - if (WARN_ON(!IS_ERR(s->context) || + if (WARN_ON(amdtp_out_stream_running(s) || (!s->pcm_channels && !s->midi_ports))) { err = -EBADFD; goto err_unlock; @@ -577,7 +577,7 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s) { mutex_lock(&s->mutex); - if (IS_ERR(s->context)) { + if (!amdtp_out_stream_running(s)) { mutex_unlock(&s->mutex); return; } diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index fd4ce3030279..28b1bf5891e6 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -1,6 +1,7 @@ #ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED #define SOUND_FIREWIRE_AMDTP_H_INCLUDED +#include #include #include #include "packets-buffer.h" @@ -92,6 +93,11 @@ void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s); unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s); void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s); +static inline bool amdtp_out_stream_running(struct amdtp_out_stream *s) +{ + return !IS_ERR(s->context); +} + /** * amdtp_out_stream_set_pcm - configure format of PCM samples * @s: the AMDTP output stream to be configured diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index ef04089bde7f..3591aebd1bb7 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -246,7 +246,6 @@ struct dice { int dev_lock_count; /* > 0 driver, < 0 userspace */ bool dev_lock_changed; bool global_enabled; - bool stream_running; wait_queue_head_t hwdep_wait; u32 notification_bits; struct snd_pcm_substream *pcm; @@ -654,7 +653,7 @@ static int dice_stream_start_packets(struct dice *dice) { int err; - if (dice->stream_running) + if (amdtp_out_stream_running(&dice->stream)) return 0; err = amdtp_out_stream_start(&dice->stream, dice->resources.channel, @@ -668,8 +667,6 @@ static int dice_stream_start_packets(struct dice *dice) return err; } - dice->stream_running = true; - return 0; } @@ -712,14 +709,10 @@ error: static void dice_stream_stop_packets(struct dice *dice) { - if (!dice->stream_running) - return; - - dice_enable_clear(dice); - - amdtp_out_stream_stop(&dice->stream); - - dice->stream_running = false; + if (amdtp_out_stream_running(&dice->stream)) { + dice_enable_clear(dice); + amdtp_out_stream_stop(&dice->stream); + } } static void dice_stream_stop(struct dice *dice) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 2c6386503940..0ac56304baff 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -53,7 +53,6 @@ struct fwspk { struct mutex mutex; struct cmp_connection connection; struct amdtp_out_stream stream; - bool stream_running; bool mute; s16 volume[6]; s16 volume_min; @@ -189,10 +188,9 @@ static int fwspk_close(struct snd_pcm_substream *substream) static void fwspk_stop_stream(struct fwspk *fwspk) { - if (fwspk->stream_running) { + if (amdtp_out_stream_running(&fwspk->stream)) { amdtp_out_stream_stop(&fwspk->stream); cmp_connection_break(&fwspk->connection); - fwspk->stream_running = false; } } @@ -286,7 +284,7 @@ static int fwspk_prepare(struct snd_pcm_substream *substream) if (amdtp_out_streaming_error(&fwspk->stream)) fwspk_stop_stream(fwspk); - if (!fwspk->stream_running) { + if (!amdtp_out_stream_running(&fwspk->stream)) { err = cmp_connection_establish(&fwspk->connection, amdtp_out_stream_get_max_payload(&fwspk->stream)); if (err < 0) @@ -297,8 +295,6 @@ static int fwspk_prepare(struct snd_pcm_substream *substream) fwspk->connection.speed); if (err < 0) goto err_connection; - - fwspk->stream_running = true; } mutex_unlock(&fwspk->mutex); -- cgit From 54e72f0ba31e1562e6d6277d0e4d39a7004f814d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:15:54 +0200 Subject: ALSA: dice: reorganize interface definitions Move the DICE interface symbols into a separate header file, and add more documentation. Signed-off-by: Clemens Ladisch --- sound/firewire/dice-interface.h | 371 ++++++++++++++++++++++++++++++++++++++++ sound/firewire/dice.c | 209 +--------------------- 2 files changed, 373 insertions(+), 207 deletions(-) create mode 100644 sound/firewire/dice-interface.h diff --git a/sound/firewire/dice-interface.h b/sound/firewire/dice-interface.h new file mode 100644 index 000000000000..af916b9d07a4 --- /dev/null +++ b/sound/firewire/dice-interface.h @@ -0,0 +1,371 @@ +#ifndef SOUND_FIREWIRE_DICE_INTERFACE_H_INCLUDED +#define SOUND_FIREWIRE_DICE_INTERFACE_H_INCLUDED + +/* + * DICE device interface definitions + */ + +/* + * Generally, all registers can be read like memory, i.e., with quadlet read or + * block read transactions with any alignment or length. Writes are not + * allowed except where noted; quadlet-sized registers must be written with + * a quadlet write transaction. + * + * All values are in big endian. The DICE firmware runs on a little-endian CPU + * and just byte-swaps _all_ quadlets on the bus, so values without endianness + * (e.g. strings) get scrambled and must be byte-swapped again by the driver. + */ + +/* + * Streaming is handled by the "DICE driver" interface. Its registers are + * located in this private address space. + */ +#define DICE_PRIVATE_SPACE 0xffffe0000000uLL + +/* + * The registers are organized in several sections, which are organized + * separately to allow them to be extended individually. Whether a register is + * supported can be detected by checking its offset against its section's size. + * + * The section offset values are relative to DICE_PRIVATE_SPACE; the offset/ + * size values are measured in quadlets. Read-only. + */ +#define DICE_GLOBAL_OFFSET 0x00 +#define DICE_GLOBAL_SIZE 0x04 +#define DICE_TX_OFFSET 0x08 +#define DICE_TX_SIZE 0x0c +#define DICE_RX_OFFSET 0x10 +#define DICE_RX_SIZE 0x14 +#define DICE_EXT_SYNC_OFFSET 0x18 +#define DICE_EXT_SYNC_SIZE 0x1c +#define DICE_UNUSED2_OFFSET 0x20 +#define DICE_UNUSED2_SIZE 0x24 + +/* + * Global settings. + */ + +/* + * Stores the full 64-bit address (node ID and offset in the node's address + * space) where the device will send notifications. Must be changed with + * a compare/swap transaction by the owner. This register is automatically + * cleared on a bus reset. + */ +#define GLOBAL_OWNER 0x000 +#define OWNER_NO_OWNER 0xffff000000000000uLL +#define OWNER_NODE_SHIFT 48 + +/* + * A bitmask with asynchronous events; read-only. When any event(s) happen, + * the bits of previous events are cleared, and the value of this register is + * also written to the address stored in the owner register. + */ +#define GLOBAL_NOTIFICATION 0x008 +/* Some registers in the Rx/Tx sections may have changed. */ +#define NOTIFY_RX_CFG_CHG 0x00000001 +#define NOTIFY_TX_CFG_CHG 0x00000002 +/* Lock status of the current clock source may have changed. */ +#define NOTIFY_LOCK_CHG 0x00000010 +/* Write to the clock select register has been finished. */ +#define NOTIFY_CLOCK_ACCEPTED 0x00000020 +/* Lock status of some clock source has changed. */ +#define NOTIFY_EXT_STATUS 0x00000040 +/* Other bits may be used for device-specific events. */ + +/* + * A name that can be customized for each device; read/write. Padded with zero + * bytes. Quadlets are byte-swapped. The encoding is whatever the host driver + * happens to be using. + */ +#define GLOBAL_NICK_NAME 0x00c +#define NICK_NAME_SIZE 64 + +/* + * The current sample rate and clock source; read/write. Whether a clock + * source or sample rate is supported is device-specific; the internal clock + * source is always available. Low/mid/high = up to 48/96/192 kHz. This + * register can be changed even while streams are running. + */ +#define GLOBAL_CLOCK_SELECT 0x04c +#define CLOCK_SOURCE_MASK 0x000000ff +#define CLOCK_SOURCE_AES1 0x00000000 +#define CLOCK_SOURCE_AES2 0x00000001 +#define CLOCK_SOURCE_AES3 0x00000002 +#define CLOCK_SOURCE_AES4 0x00000003 +#define CLOCK_SOURCE_AES_ANY 0x00000004 +#define CLOCK_SOURCE_ADAT 0x00000005 +#define CLOCK_SOURCE_TDIF 0x00000006 +#define CLOCK_SOURCE_WC 0x00000007 +#define CLOCK_SOURCE_ARX1 0x00000008 +#define CLOCK_SOURCE_ARX2 0x00000009 +#define CLOCK_SOURCE_ARX3 0x0000000a +#define CLOCK_SOURCE_ARX4 0x0000000b +#define CLOCK_SOURCE_INTERNAL 0x0000000c +#define CLOCK_RATE_MASK 0x0000ff00 +#define CLOCK_RATE_32000 0x00000000 +#define CLOCK_RATE_44100 0x00000100 +#define CLOCK_RATE_48000 0x00000200 +#define CLOCK_RATE_88200 0x00000300 +#define CLOCK_RATE_96000 0x00000400 +#define CLOCK_RATE_176400 0x00000500 +#define CLOCK_RATE_192000 0x00000600 +#define CLOCK_RATE_ANY_LOW 0x00000700 +#define CLOCK_RATE_ANY_MID 0x00000800 +#define CLOCK_RATE_ANY_HIGH 0x00000900 +#define CLOCK_RATE_NONE 0x00000a00 +#define CLOCK_RATE_SHIFT 8 + +/* + * Enable streaming; read/write. Writing a non-zero value (re)starts all + * streams that have a valid iso channel set; zero stops all streams. The + * streams' parameters must be configured before starting. This register is + * automatically cleared on a bus reset. + */ +#define GLOBAL_ENABLE 0x050 + +/* + * Status of the sample clock; read-only. + */ +#define GLOBAL_STATUS 0x054 +/* The current clock source is locked. */ +#define STATUS_SOURCE_LOCKED 0x00000001 +/* The actual sample rate; CLOCK_RATE_32000-_192000 or _NONE. */ +#define STATUS_NOMINAL_RATE_MASK 0x0000ff00 + +/* + * Status of all clock sources; read-only. + */ +#define GLOBAL_EXTENDED_STATUS 0x058 +/* + * The _LOCKED bits always show the current status; any change generates + * a notification. + */ +#define EXT_STATUS_AES1_LOCKED 0x00000001 +#define EXT_STATUS_AES2_LOCKED 0x00000002 +#define EXT_STATUS_AES3_LOCKED 0x00000004 +#define EXT_STATUS_AES4_LOCKED 0x00000008 +#define EXT_STATUS_ADAT_LOCKED 0x00000010 +#define EXT_STATUS_TDIF_LOCKED 0x00000020 +#define EXT_STATUS_ARX1_LOCKED 0x00000040 +#define EXT_STATUS_ARX2_LOCKED 0x00000080 +#define EXT_STATUS_ARX3_LOCKED 0x00000100 +#define EXT_STATUS_ARX4_LOCKED 0x00000200 +#define EXT_STATUS_WC_LOCKED 0x00000400 +/* + * The _SLIP bits do not generate notifications; a set bit indicates that an + * error occurred since the last time when this register was read with + * a quadlet read transaction. + */ +#define EXT_STATUS_AES1_SLIP 0x00010000 +#define EXT_STATUS_AES2_SLIP 0x00020000 +#define EXT_STATUS_AES3_SLIP 0x00040000 +#define EXT_STATUS_AES4_SLIP 0x00080000 +#define EXT_STATUS_ADAT_SLIP 0x00100000 +#define EXT_STATUS_TDIF_SLIP 0x00200000 +#define EXT_STATUS_ARX1_SLIP 0x00400000 +#define EXT_STATUS_ARX2_SLIP 0x00800000 +#define EXT_STATUS_ARX3_SLIP 0x01000000 +#define EXT_STATUS_ARX4_SLIP 0x02000000 +#define EXT_STATUS_WC_SLIP 0x04000000 + +/* + * The measured rate of the current clock source, in Hz; read-only. + */ +#define GLOBAL_SAMPLE_RATE 0x05c + +/* + * The version of the DICE driver specification that this device conforms to; + * read-only. + */ +#define GLOBAL_VERSION 0x060 + +/* Some old firmware versions do not have the following global registers: */ + +/* + * Supported sample rates and clock sources; read-only. + */ +#define GLOBAL_CLOCK_CAPABILITIES 0x064 +#define CLOCK_CAP_RATE_32000 0x00000001 +#define CLOCK_CAP_RATE_44100 0x00000002 +#define CLOCK_CAP_RATE_48000 0x00000004 +#define CLOCK_CAP_RATE_88200 0x00000008 +#define CLOCK_CAP_RATE_96000 0x00000010 +#define CLOCK_CAP_RATE_176400 0x00000020 +#define CLOCK_CAP_RATE_192000 0x00000040 +#define CLOCK_CAP_SOURCE_AES1 0x00010000 +#define CLOCK_CAP_SOURCE_AES2 0x00020000 +#define CLOCK_CAP_SOURCE_AES3 0x00040000 +#define CLOCK_CAP_SOURCE_AES4 0x00080000 +#define CLOCK_CAP_SOURCE_AES_ANY 0x00100000 +#define CLOCK_CAP_SOURCE_ADAT 0x00200000 +#define CLOCK_CAP_SOURCE_TDIF 0x00400000 +#define CLOCK_CAP_SOURCE_WC 0x00800000 +#define CLOCK_CAP_SOURCE_ARX1 0x01000000 +#define CLOCK_CAP_SOURCE_ARX2 0x02000000 +#define CLOCK_CAP_SOURCE_ARX3 0x04000000 +#define CLOCK_CAP_SOURCE_ARX4 0x08000000 +#define CLOCK_CAP_SOURCE_INTERNAL 0x10000000 + +/* + * Names of all clock sources; read-only. Quadlets are byte-swapped. Names + * are separated with one backslash, the list is terminated with two + * backslashes. Unused clock sources are included. + */ +#define GLOBAL_CLOCK_SOURCE_NAMES 0x068 +#define CLOCK_SOURCE_NAMES_SIZE 256 + +/* + * Capture stream settings. This section includes the number/size registers + * and the registers of all streams. + */ + +/* + * The number of supported capture streams; read-only. + */ +#define TX_NUMBER 0x000 + +/* + * The size of one stream's register block, in quadlets; read-only. The + * registers of the first stream follow immediately afterwards; the registers + * of the following streams are offset by this register's value. + */ +#define TX_SIZE 0x004 + +/* + * The isochronous channel number on which packets are sent, or -1 if the + * stream is not to be used; read/write. + */ +#define TX_ISOCHRONOUS 0x008 + +/* + * The number of audio channels; read-only. There will be one quadlet per + * channel; the first channel is the first quadlet in a data block. + */ +#define TX_NUMBER_AUDIO 0x00c + +/* + * The number of MIDI ports, 0-8; read-only. If > 0, there will be one + * additional quadlet in each data block, following the audio quadlets. + */ +#define TX_NUMBER_MIDI 0x010 + +/* + * The speed at which the packets are sent, SCODE_100-_400; read/write. + */ +#define TX_SPEED 0x014 + +/* + * Names of all audio channels; read-only. Quadlets are byte-swapped. Names + * are separated with one backslash, the list is terminated with two + * backslashes. + */ +#define TX_NAMES 0x018 +#define TX_NAMES_SIZE 256 + +/* + * Audio IEC60958 capabilities; read-only. Bitmask with one bit per audio + * channel. + */ +#define TX_AC3_CAPABILITIES 0x118 + +/* + * Send audio data with IEC60958 label; read/write. Bitmask with one bit per + * audio channel. This register can be changed even while the stream is + * running. + */ +#define TX_AC3_ENABLE 0x11c + +/* + * Playback stream settings. This section includes the number/size registers + * and the registers of all streams. + */ + +/* + * The number of supported playback streams; read-only. + */ +#define RX_NUMBER 0x000 + +/* + * The size of one stream's register block, in quadlets; read-only. The + * registers of the first stream follow immediately afterwards; the registers + * of the following streams are offset by this register's value. + */ +#define RX_SIZE 0x004 + +/* + * The isochronous channel number on which packets are received, or -1 if the + * stream is not to be used; read/write. + */ +#define RX_ISOCHRONOUS 0x008 + +/* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x00c + +/* + * The number of audio channels; read-only. There will be one quadlet per + * channel. + */ +#define RX_NUMBER_AUDIO 0x010 + +/* + * The number of MIDI ports, 0-8; read-only. If > 0, there will be one + * additional quadlet in each data block, following the audio quadlets. + */ +#define RX_NUMBER_MIDI 0x014 + +/* + * Names of all audio channels; read-only. Quadlets are byte-swapped. Names + * are separated with one backslash, the list is terminated with two + * backslashes. + */ +#define RX_NAMES 0x018 +#define RX_NAMES_SIZE 256 + +/* + * Audio IEC60958 capabilities; read-only. Bitmask with one bit per audio + * channel. + */ +#define RX_AC3_CAPABILITIES 0x118 + +/* + * Receive audio data with IEC60958 label; read/write. Bitmask with one bit + * per audio channel. This register can be changed even while the stream is + * running. + */ +#define RX_AC3_ENABLE 0x11c + +/* + * Extended synchronization information. + * This section can be read completely with a block read request. + */ + +/* + * Current clock source; read-only. + */ +#define EXT_SYNC_CLOCK_SOURCE 0x000 + +/* + * Clock source is locked (boolean); read-only. + */ +#define EXT_SYNC_LOCKED 0x004 + +/* + * Current sample rate (CLOCK_RATE_* >> CLOCK_RATE_SHIFT), _32000-_192000 or + * _NONE; read-only. + */ +#define EXT_SYNC_RATE 0x008 + +/* + * ADAT user data bits; read-only. + */ +#define EXT_SYNC_ADAT_USER_DATA 0x00c +/* The data bits, if available. */ +#define ADAT_USER_DATA_MASK 0x0f +/* The data bits are not available. */ +#define ADAT_USER_DATA_NO_DATA 0x10 + +#endif diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 3591aebd1bb7..1da1ddefd6a8 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -26,212 +26,7 @@ #include "amdtp.h" #include "iso-resources.h" #include "lib.h" - -#define DICE_PRIVATE_SPACE 0xffffe0000000uLL - -/* offset from DICE_PRIVATE_SPACE; offsets and sizes in quadlets */ -#define DICE_GLOBAL_OFFSET 0x00 -#define DICE_GLOBAL_SIZE 0x04 -#define DICE_TX_OFFSET 0x08 -#define DICE_TX_SIZE 0x0c -#define DICE_RX_OFFSET 0x10 -#define DICE_RX_SIZE 0x14 - -/* pointed to by DICE_GLOBAL_OFFSET */ -#define GLOBAL_OWNER 0x000 -#define OWNER_NO_OWNER 0xffff000000000000uLL -#define OWNER_NODE_SHIFT 48 -#define GLOBAL_NOTIFICATION 0x008 -#define NOTIFY_RX_CFG_CHG 0x00000001 -#define NOTIFY_TX_CFG_CHG 0x00000002 -#define NOTIFY_DUP_ISOC 0x00000004 -#define NOTIFY_BW_ERR 0x00000008 -#define NOTIFY_LOCK_CHG 0x00000010 -#define NOTIFY_CLOCK_ACCEPTED 0x00000020 -#define NOTIFY_INTERFACE_CHG 0x00000040 -#define NOTIFY_MESSAGE 0x00100000 -#define GLOBAL_NICK_NAME 0x00c -#define NICK_NAME_SIZE 64 -#define GLOBAL_CLOCK_SELECT 0x04c -#define CLOCK_SOURCE_MASK 0x000000ff -#define CLOCK_SOURCE_AES1 0x00000000 -#define CLOCK_SOURCE_AES2 0x00000001 -#define CLOCK_SOURCE_AES3 0x00000002 -#define CLOCK_SOURCE_AES4 0x00000003 -#define CLOCK_SOURCE_AES_ANY 0x00000004 -#define CLOCK_SOURCE_ADAT 0x00000005 -#define CLOCK_SOURCE_TDIF 0x00000006 -#define CLOCK_SOURCE_WC 0x00000007 -#define CLOCK_SOURCE_ARX1 0x00000008 -#define CLOCK_SOURCE_ARX2 0x00000009 -#define CLOCK_SOURCE_ARX3 0x0000000a -#define CLOCK_SOURCE_ARX4 0x0000000b -#define CLOCK_SOURCE_INTERNAL 0x0000000c -#define CLOCK_RATE_MASK 0x0000ff00 -#define CLOCK_RATE_32000 0x00000000 -#define CLOCK_RATE_44100 0x00000100 -#define CLOCK_RATE_48000 0x00000200 -#define CLOCK_RATE_88200 0x00000300 -#define CLOCK_RATE_96000 0x00000400 -#define CLOCK_RATE_176400 0x00000500 -#define CLOCK_RATE_192000 0x00000600 -#define CLOCK_RATE_ANY_LOW 0x00000700 -#define CLOCK_RATE_ANY_MID 0x00000800 -#define CLOCK_RATE_ANY_HIGH 0x00000900 -#define CLOCK_RATE_NONE 0x00000a00 -#define CLOCK_RATE_SHIFT 8 -#define GLOBAL_ENABLE 0x050 -#define ENABLE 0x00000001 -#define GLOBAL_STATUS 0x054 -#define STATUS_SOURCE_LOCKED 0x00000001 -#define STATUS_RATE_CONFLICT 0x00000002 -#define STATUS_NOMINAL_RATE_MASK 0x0000ff00 -#define GLOBAL_EXTENDED_STATUS 0x058 -#define EXT_STATUS_AES1_LOCKED 0x00000001 -#define EXT_STATUS_AES2_LOCKED 0x00000002 -#define EXT_STATUS_AES3_LOCKED 0x00000004 -#define EXT_STATUS_AES4_LOCKED 0x00000008 -#define EXT_STATUS_ADAT_LOCKED 0x00000010 -#define EXT_STATUS_TDIF_LOCKED 0x00000020 -#define EXT_STATUS_ARX1_LOCKED 0x00000040 -#define EXT_STATUS_ARX2_LOCKED 0x00000080 -#define EXT_STATUS_ARX3_LOCKED 0x00000100 -#define EXT_STATUS_ARX4_LOCKED 0x00000200 -#define EXT_STATUS_WC_LOCKED 0x00000400 -#define EXT_STATUS_AES1_SLIP 0x00010000 -#define EXT_STATUS_AES2_SLIP 0x00020000 -#define EXT_STATUS_AES3_SLIP 0x00040000 -#define EXT_STATUS_AES4_SLIP 0x00080000 -#define EXT_STATUS_ADAT_SLIP 0x00100000 -#define EXT_STATUS_TDIF_SLIP 0x00200000 -#define EXT_STATUS_ARX1_SLIP 0x00400000 -#define EXT_STATUS_ARX2_SLIP 0x00800000 -#define EXT_STATUS_ARX3_SLIP 0x01000000 -#define EXT_STATUS_ARX4_SLIP 0x02000000 -#define EXT_STATUS_WC_SLIP 0x04000000 -#define GLOBAL_SAMPLE_RATE 0x05c -#define GLOBAL_VERSION 0x060 -#define GLOBAL_CLOCK_CAPABILITIES 0x064 -#define CLOCK_CAP_RATE_32000 0x00000001 -#define CLOCK_CAP_RATE_44100 0x00000002 -#define CLOCK_CAP_RATE_48000 0x00000004 -#define CLOCK_CAP_RATE_88200 0x00000008 -#define CLOCK_CAP_RATE_96000 0x00000010 -#define CLOCK_CAP_RATE_176400 0x00000020 -#define CLOCK_CAP_RATE_192000 0x00000040 -#define CLOCK_CAP_SOURCE_AES1 0x00010000 -#define CLOCK_CAP_SOURCE_AES2 0x00020000 -#define CLOCK_CAP_SOURCE_AES3 0x00040000 -#define CLOCK_CAP_SOURCE_AES4 0x00080000 -#define CLOCK_CAP_SOURCE_AES_ANY 0x00100000 -#define CLOCK_CAP_SOURCE_ADAT 0x00200000 -#define CLOCK_CAP_SOURCE_TDIF 0x00400000 -#define CLOCK_CAP_SOURCE_WC 0x00800000 -#define CLOCK_CAP_SOURCE_ARX1 0x01000000 -#define CLOCK_CAP_SOURCE_ARX2 0x02000000 -#define CLOCK_CAP_SOURCE_ARX3 0x04000000 -#define CLOCK_CAP_SOURCE_ARX4 0x08000000 -#define CLOCK_CAP_SOURCE_INTERNAL 0x10000000 -#define GLOBAL_CLOCK_SOURCE_NAMES 0x068 -#define CLOCK_SOURCE_NAMES_SIZE 256 - -/* pointed to by DICE_TX_OFFSET */ -#define TX_NUMBER 0x000 -#define TX_SIZE 0x004 -/* repeated TX_NUMBER times, offset by TX_SIZE quadlets */ -#define TX_ISOCHRONOUS 0x008 -#define TX_NUMBER_AUDIO 0x00c -#define TX_NUMBER_MIDI 0x010 -#define TX_SPEED 0x014 -#define TX_NAMES 0x018 -#define TX_NAMES_SIZE 256 -#define TX_AC3_CAPABILITIES 0x118 -#define TX_AC3_ENABLE 0x11c - -/* pointed to by DICE_RX_OFFSET */ -#define RX_NUMBER 0x000 -#define RX_SIZE 0x004 -/* repeated RX_NUMBER times, offset by RX_SIZE quadlets */ -#define RX_ISOCHRONOUS 0x008 -#define RX_SEQ_START 0x00c -#define RX_NUMBER_AUDIO 0x010 -#define RX_NUMBER_MIDI 0x014 -#define RX_NAMES 0x018 -#define RX_NAMES_SIZE 256 -#define RX_AC3_CAPABILITIES 0x118 -#define RX_AC3_ENABLE 0x11c - - -#define FIRMWARE_LOAD_SPACE 0xffffe0100000uLL - -/* offset from FIRMWARE_LOAD_SPACE */ -#define FIRMWARE_VERSION 0x000 -#define FIRMWARE_OPCODE 0x004 -#define OPCODE_MASK 0x00000fff -#define OPCODE_GET_IMAGE_DESC 0x00000000 -#define OPCODE_DELETE_IMAGE 0x00000001 -#define OPCODE_CREATE_IMAGE 0x00000002 -#define OPCODE_UPLOAD 0x00000003 -#define OPCODE_UPLOAD_STAT 0x00000004 -#define OPCODE_RESET_IMAGE 0x00000005 -#define OPCODE_TEST_ACTION 0x00000006 -#define OPCODE_GET_RUNNING_IMAGE_VINFO 0x0000000a -#define OPCODE_EXECUTE 0x80000000 -#define FIRMWARE_RETURN_STATUS 0x008 -#define FIRMWARE_PROGRESS 0x00c -#define PROGRESS_CURR_MASK 0x00000fff -#define PROGRESS_MAX_MASK 0x00fff000 -#define PROGRESS_TOUT_MASK 0x0f000000 -#define PROGRESS_FLAG 0x80000000 -#define FIRMWARE_CAPABILITIES 0x010 -#define FL_CAP_AUTOERASE 0x00000001 -#define FL_CAP_PROGRESS 0x00000002 -#define FIRMWARE_DATA 0x02c -#define TEST_CMD_POKE 0x00000001 -#define TEST_CMD_PEEK 0x00000002 -#define CMD_GET_AVS_CNT 0x00000003 -#define CMD_CLR_AVS_CNT 0x00000004 -#define CMD_SET_MODE 0x00000005 -#define CMD_SET_MIDIBP 0x00000006 -#define CMD_GET_AVSPHASE 0x00000007 -#define CMD_ENABLE_BNC_SYNC 0x00000008 -#define CMD_PULSE_BNC_SYNC 0x00000009 -#define CMD_EMUL_SLOW_CMD 0x0000000a -#define FIRMWARE_TEST_DELAY 0xfd8 -#define FIRMWARE_TEST_BUF 0xfdc - - -/* EAP */ -#define EAP_PRIVATE_SPACE 0xffffe0200000uLL - -#define EAP_CAPABILITY_OFFSET 0x000 -#define EAP_CAPABILITY_SIZE 0x004 -/* ... */ - -#define EAP_ROUTER_CAPS 0x000 -#define ROUTER_EXPOSED 0x00000001 -#define ROUTER_READ_ONLY 0x00000002 -#define ROUTER_FLASH 0x00000004 -#define MAX_ROUTES_MASK 0xffff0000 -#define EAP_MIXER_CAPS 0x004 -#define MIXER_EXPOSED 0x00000001 -#define MIXER_READ_ONLY 0x00000002 -#define MIXER_FLASH 0x00000004 -#define MIXER_IN_DEV_MASK 0x000000f0 -#define MIXER_OUT_DEV_MASK 0x00000f00 -#define MIXER_INPUTS_MASK 0x00ff0000 -#define MIXER_OUTPUTS_MASK 0xff000000 -#define EAP_GENERAL_CAPS 0x008 -#define GENERAL_STREAM_CONFIG 0x00000001 -#define GENERAL_FLASH 0x00000002 -#define GENERAL_PEAK 0x00000004 -#define GENERAL_MAX_TX_STREAMS_MASK 0x000000f0 -#define GENERAL_MAX_RX_STREAMS_MASK 0x00000f00 -#define GENERAL_STREAM_CONFIG_FLASH 0x00001000 -#define GENERAL_CHIP_MASK 0x00ff0000 -#define GENERAL_CHIP_DICE_II 0x00000000 -#define GENERAL_CHIP_DICE_MINI 0x00010000 -#define GENERAL_CHIP_DICE_JR 0x00020000 +#include "dice-interface.h" struct dice { @@ -479,7 +274,7 @@ static int dice_enable_set(struct dice *dice) __be32 value; int rcode, err, errors = 0; - value = cpu_to_be32(ENABLE); + value = cpu_to_be32(1); for (;;) { rcode = fw_run_transaction(device->card, TCODE_WRITE_QUADLET_REQUEST, -- cgit From cbab328ddc78589233be8be2f1e6a5f9d97b81db Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:16:02 +0200 Subject: ALSA: dice: fix device detection for other vendors DICE devices do not have a unique specifier ID in their unit directory (it's always the same as the device vendor's ID), so rely on just the version ID for driver loading, and use a heuristic in the probe callback to detect actual DICE devices. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 102 +++++++++++++++++++++++++++++++++++++++++--------- 1 file changed, 84 insertions(+), 18 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 1da1ddefd6a8..b4827ff45d68 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -827,28 +827,93 @@ static void dice_card_free(struct snd_card *card) mutex_destroy(&dice->mutex); } +#define DICE_CATEGORY_ID 0x04 + +static int dice_interface_check(struct fw_unit *unit) +{ + static const int min_values[10] = { + 10, 0x64 / 4, + 10, 0x18 / 4, + 10, 0x18 / 4, + 0, 0, + 0, 0, + }; + struct fw_device *device = fw_parent_device(unit); + struct fw_csr_iterator it; + int key, value, vendor = -1, model = -1, err; + unsigned int i; + __be32 pointers[ARRAY_SIZE(min_values)]; + __be32 version; + + /* + * Check that GUID and unit directory are constructed according to DICE + * rules, i.e., that the specifier ID is the GUID's OUI, and that the + * GUID chip ID consists of the 8-bit DICE category ID, the 10-bit + * product ID, and a 22-bit serial number. + */ + fw_csr_iterator_init(&it, unit->directory); + while (fw_csr_iterator_next(&it, &key, &value)) { + switch (key) { + case CSR_SPECIFIER_ID: + vendor = value; + break; + case CSR_MODEL: + model = value; + break; + } + } + if (device->config_rom[3] != ((vendor << 8) | DICE_CATEGORY_ID) || + device->config_rom[4] >> 22 != model) + return -ENODEV; + + /* + * Check that the sub address spaces exist and are located inside the + * private address space. The minimum values are chosen so that all + * minimally required registers are included. + */ + err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE, + pointers, sizeof(pointers)); + if (err < 0) + return -ENODEV; + for (i = 0; i < ARRAY_SIZE(pointers); ++i) { + value = be32_to_cpu(pointers[i]); + if (value < min_values[i] || value >= 0x40000) + return -ENODEV; + } + + /* + * Check that the implemented DICE driver specification major version + * number matches. + */ + err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, + &version, 4); + if (err < 0) + return -ENODEV; + if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { + dev_err(&unit->device, + "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); + return -ENODEV; + } + + return 0; +} + static int dice_init_offsets(struct dice *dice) { __be32 pointers[6]; - unsigned int global_size, rx_size; int err; err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE, &pointers, 6 * 4); + DICE_PRIVATE_SPACE, + pointers, sizeof(pointers)); if (err < 0) return err; dice->global_offset = be32_to_cpu(pointers[0]) * 4; - global_size = be32_to_cpu(pointers[1]); dice->rx_offset = be32_to_cpu(pointers[4]) * 4; - rx_size = be32_to_cpu(pointers[5]); - - /* some sanity checks to ensure that we actually have a DICE */ - if (dice->global_offset < 10 * 4 || global_size < 0x168 / 4 || - dice->rx_offset < 10 * 4 || rx_size < 0x120 / 4) { - dev_err(&dice->unit->device, "invalid register pointers\n"); - return -ENXIO; - } return 0; } @@ -881,8 +946,8 @@ static void dice_card_strings(struct dice *dice) strcpy(model, "?"); fw_csr_string(dice->unit->directory, CSR_MODEL, model, sizeof(model)); snprintf(card->longname, sizeof(card->longname), - "%s %s, GUID %08x%08x at %s, S%d", - vendor, model, dev->config_rom[3], dev->config_rom[4], + "%s %s (serial %u) at %s, S%d", + vendor, model, dev->config_rom[4] & 0x3fffff, dev_name(&dice->unit->device), 100 << dev->max_speed); strcpy(card->mixername, "DICE"); @@ -895,6 +960,10 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) __be32 clock_sel; int err; + err = dice_interface_check(unit); + if (err < 0) + return err; + err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*dice), &card); if (err < 0) return err; @@ -1020,15 +1089,12 @@ static void dice_bus_reset(struct fw_unit *unit) mutex_unlock(&dice->mutex); } -#define TC_OUI 0x000166 #define DICE_INTERFACE 0x000001 static const struct ieee1394_device_id dice_id_table[] = { { - .match_flags = IEEE1394_MATCH_SPECIFIER_ID | - IEEE1394_MATCH_VERSION, - .specifier_id = TC_OUI, - .version = DICE_INTERFACE, + .match_flags = IEEE1394_MATCH_VERSION, + .version = DICE_INTERFACE, }, { } }; -- cgit From a7304e3bf0489d3fc0260bdb9c1441427a26a38f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:16:10 +0200 Subject: ALSA: dice: support dual-wire stream format at 192 kHz Change the AMDTP streaming code to handle the non-standard stream format that DICE devices use at sample rates greater than 96 kHz. Signed-off-by: Clemens Ladisch --- sound/firewire/amdtp.c | 155 +++++++++++++++++++++++++++++++++++++--------- sound/firewire/amdtp.h | 41 ++++-------- sound/firewire/dice.c | 29 ++++++--- sound/firewire/speakers.c | 6 +- 4 files changed, 164 insertions(+), 67 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index d56b8e736b7d..a09c3b393c0b 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -65,42 +65,66 @@ void amdtp_out_stream_destroy(struct amdtp_out_stream *s) } EXPORT_SYMBOL(amdtp_out_stream_destroy); +unsigned int amdtp_syt_intervals[CIP_SFC_COUNT] = { + [CIP_SFC_32000] = 8, + [CIP_SFC_44100] = 8, + [CIP_SFC_48000] = 8, + [CIP_SFC_88200] = 16, + [CIP_SFC_96000] = 16, + [CIP_SFC_176400] = 32, + [CIP_SFC_192000] = 32, +}; +EXPORT_SYMBOL(amdtp_syt_intervals); + /** - * amdtp_out_stream_set_rate - set the sample rate + * amdtp_out_stream_set_parameters - set stream parameters * @s: the AMDTP output stream to configure * @rate: the sample rate + * @pcm_channels: the number of PCM samples in each data block, to be encoded + * as AM824 multi-bit linear audio + * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels) * - * The sample rate must be set before the stream is started, and must not be + * The parameters must be set before the stream is started, and must not be * changed while the stream is running. */ -void amdtp_out_stream_set_rate(struct amdtp_out_stream *s, unsigned int rate) +void amdtp_out_stream_set_parameters(struct amdtp_out_stream *s, + unsigned int rate, + unsigned int pcm_channels, + unsigned int midi_ports) { - static const struct { - unsigned int rate; - unsigned int syt_interval; - } rate_info[] = { - [CIP_SFC_32000] = { 32000, 8, }, - [CIP_SFC_44100] = { 44100, 8, }, - [CIP_SFC_48000] = { 48000, 8, }, - [CIP_SFC_88200] = { 88200, 16, }, - [CIP_SFC_96000] = { 96000, 16, }, - [CIP_SFC_176400] = { 176400, 32, }, - [CIP_SFC_192000] = { 192000, 32, }, + static const unsigned int rates[] = { + [CIP_SFC_32000] = 32000, + [CIP_SFC_44100] = 44100, + [CIP_SFC_48000] = 48000, + [CIP_SFC_88200] = 88200, + [CIP_SFC_96000] = 96000, + [CIP_SFC_176400] = 176400, + [CIP_SFC_192000] = 192000, }; unsigned int sfc; if (WARN_ON(amdtp_out_stream_running(s))) return; - for (sfc = 0; sfc < ARRAY_SIZE(rate_info); ++sfc) - if (rate_info[sfc].rate == rate) + for (sfc = 0; sfc < CIP_SFC_COUNT; ++sfc) + if (rates[sfc] == rate) goto sfc_found; WARN_ON(1); return; sfc_found: + s->dual_wire = (s->flags & CIP_HI_DUALWIRE) && sfc > CIP_SFC_96000; + if (s->dual_wire) { + sfc -= 2; + rate /= 2; + pcm_channels *= 2; + } s->sfc = sfc; - s->syt_interval = rate_info[sfc].syt_interval; + s->data_block_quadlets = pcm_channels + DIV_ROUND_UP(midi_ports, 8); + s->pcm_channels = pcm_channels; + s->midi_ports = midi_ports; + + s->syt_interval = amdtp_syt_intervals[sfc]; /* default buffering in the device */ s->transfer_delay = TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; @@ -108,21 +132,17 @@ sfc_found: /* additional buffering needed to adjust for no-data packets */ s->transfer_delay += TICKS_PER_SECOND * s->syt_interval / rate; } -EXPORT_SYMBOL(amdtp_out_stream_set_rate); +EXPORT_SYMBOL(amdtp_out_stream_set_parameters); /** * amdtp_out_stream_get_max_payload - get the stream's packet size * @s: the AMDTP output stream * * This function must not be called before the stream has been configured - * with amdtp_out_stream_set_rate(), amdtp_out_stream_set_pcm(), and - * amdtp_out_stream_set_midi(). + * with amdtp_out_stream_set_parameters(). */ unsigned int amdtp_out_stream_get_max_payload(struct amdtp_out_stream *s) { - s->data_block_quadlets = s->pcm_channels; - s->data_block_quadlets += DIV_ROUND_UP(s->midi_ports, 8); - return 8 + s->syt_interval * s->data_block_quadlets * 4; } EXPORT_SYMBOL(amdtp_out_stream_get_max_payload); @@ -133,14 +153,21 @@ static void amdtp_write_s16(struct amdtp_out_stream *s, static void amdtp_write_s32(struct amdtp_out_stream *s, struct snd_pcm_substream *pcm, __be32 *buffer, unsigned int frames); +static void amdtp_write_s16_dualwire(struct amdtp_out_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames); +static void amdtp_write_s32_dualwire(struct amdtp_out_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames); /** * amdtp_out_stream_set_pcm_format - set the PCM format * @s: the AMDTP output stream to configure * @format: the format of the ALSA PCM device * - * The sample format must be set before the stream is started, and must not be - * changed while the stream is running. + * The sample format must be set after the other paramters (rate/PCM channels/ + * MIDI) and before the stream is started, and must not be changed while the + * stream is running. */ void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s, snd_pcm_format_t format) @@ -153,10 +180,16 @@ void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s, WARN_ON(1); /* fall through */ case SNDRV_PCM_FORMAT_S16: - s->transfer_samples = amdtp_write_s16; + if (s->dual_wire) + s->transfer_samples = amdtp_write_s16_dualwire; + else + s->transfer_samples = amdtp_write_s16; break; case SNDRV_PCM_FORMAT_S32: - s->transfer_samples = amdtp_write_s32; + if (s->dual_wire) + s->transfer_samples = amdtp_write_s32_dualwire; + else + s->transfer_samples = amdtp_write_s32; break; } } @@ -305,6 +338,68 @@ static void amdtp_write_s16(struct amdtp_out_stream *s, } } +static void amdtp_write_s32_dualwire(struct amdtp_out_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, frame_adjust_1, frame_adjust_2, i, c; + const u32 *src; + + channels = s->pcm_channels; + src = (void *)runtime->dma_area + + s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frame_adjust_1 = channels - 1; + frame_adjust_2 = 1 - (s->data_block_quadlets - channels); + + channels /= 2; + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + *buffer = cpu_to_be32((*src >> 8) | 0x40000000); + src++; + buffer += 2; + } + buffer -= frame_adjust_1; + for (c = 0; c < channels; ++c) { + *buffer = cpu_to_be32((*src >> 8) | 0x40000000); + src++; + buffer += 2; + } + buffer -= frame_adjust_2; + } +} + +static void amdtp_write_s16_dualwire(struct amdtp_out_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, frame_adjust_1, frame_adjust_2, i, c; + const u16 *src; + + channels = s->pcm_channels; + src = (void *)runtime->dma_area + + s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frame_adjust_1 = channels - 1; + frame_adjust_2 = 1 - (s->data_block_quadlets - channels); + + channels /= 2; + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + *buffer = cpu_to_be32((*src << 8) | 0x40000000); + src++; + buffer += 2; + } + buffer -= frame_adjust_1; + for (c = 0; c < channels; ++c) { + *buffer = cpu_to_be32((*src << 8) | 0x40000000); + src++; + buffer += 2; + } + buffer -= frame_adjust_2; + } +} + static void amdtp_fill_pcm_silence(struct amdtp_out_stream *s, __be32 *buffer, unsigned int frames) { @@ -390,6 +485,9 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle) s->packet_index = index; if (pcm) { + if (s->dual_wire) + data_blocks *= 2; + ptr = s->pcm_buffer_pointer + data_blocks; if (ptr >= pcm->runtime->buffer_size) ptr -= pcm->runtime->buffer_size; @@ -459,8 +557,7 @@ static int queue_initial_skip_packets(struct amdtp_out_stream *s) * @speed: firewire speed code * * The stream cannot be started until it has been configured with - * amdtp_out_stream_set_rate(), amdtp_out_stream_set_pcm(), - * amdtp_out_stream_set_midi(), and amdtp_out_stream_set_format(); + * amdtp_out_stream_set_parameters() and amdtp_out_stream_set_pcm_format(), * and it must be started before any PCM or MIDI device can be started. */ int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 28b1bf5891e6..f3d03dd92c39 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -15,10 +15,15 @@ * @CIP_BLOCKING: In blocking mode, each packet contains either zero or * SYT_INTERVAL samples, with these two types alternating so that * the overall sample rate comes out right. + * @CIP_HI_DUALWIRE: At rates above 96 kHz, pretend that the stream runs + * at half the actual sample rate with twice the number of channels; + * two samples of a channel are stored consecutively in the packet. + * Requires blocking mode and SYT_INTERVAL-aligned PCM buffer size. */ enum cip_out_flags { CIP_NONBLOCKING = 0x00, CIP_BLOCKING = 0x01, + CIP_HI_DUALWIRE = 0x02, }; /** @@ -32,6 +37,7 @@ enum cip_sfc { CIP_SFC_96000 = 4, CIP_SFC_176400 = 5, CIP_SFC_192000 = 6, + CIP_SFC_COUNT }; #define AMDTP_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \ @@ -48,6 +54,7 @@ struct amdtp_out_stream { struct mutex mutex; enum cip_sfc sfc; + bool dual_wire; unsigned int data_block_quadlets; unsigned int pcm_channels; unsigned int midi_ports; @@ -80,7 +87,10 @@ int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit, enum cip_out_flags flags); void amdtp_out_stream_destroy(struct amdtp_out_stream *s); -void amdtp_out_stream_set_rate(struct amdtp_out_stream *s, unsigned int rate); +void amdtp_out_stream_set_parameters(struct amdtp_out_stream *s, + unsigned int rate, + unsigned int pcm_channels, + unsigned int midi_ports); unsigned int amdtp_out_stream_get_max_payload(struct amdtp_out_stream *s); int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed); @@ -93,38 +103,13 @@ void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s); unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s); void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s); +extern unsigned int amdtp_syt_intervals[CIP_SFC_COUNT]; + static inline bool amdtp_out_stream_running(struct amdtp_out_stream *s) { return !IS_ERR(s->context); } -/** - * amdtp_out_stream_set_pcm - configure format of PCM samples - * @s: the AMDTP output stream to be configured - * @pcm_channels: the number of PCM samples in each data block, to be encoded - * as AM824 multi-bit linear audio - * - * This function must not be called while the stream is running. - */ -static inline void amdtp_out_stream_set_pcm(struct amdtp_out_stream *s, - unsigned int pcm_channels) -{ - s->pcm_channels = pcm_channels; -} - -/** - * amdtp_out_stream_set_midi - configure format of MIDI data - * @s: the AMDTP output stream to be configured - * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels) - * - * This function must not be called while the stream is running. - */ -static inline void amdtp_out_stream_set_midi(struct amdtp_out_stream *s, - unsigned int midi_ports) -{ - s->midi_ports = midi_ports; -} - /** * amdtp_out_streaming_error - check for streaming error * @s: the AMDTP output stream diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index b4827ff45d68..8804e42a96c6 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -375,7 +375,7 @@ static int dice_open(struct snd_pcm_substream *substream) struct dice *dice = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; __be32 clock_sel, number_audio, number_midi; - unsigned int rate; + unsigned int rate_index, rate; int err; err = dice_try_lock(dice); @@ -387,12 +387,13 @@ static int dice_open(struct snd_pcm_substream *substream) &clock_sel, 4); if (err < 0) goto err_lock; - rate = (be32_to_cpu(clock_sel) & CLOCK_RATE_MASK) >> CLOCK_RATE_SHIFT; - if (rate >= ARRAY_SIZE(dice_rates)) { + rate_index = (be32_to_cpu(clock_sel) & CLOCK_RATE_MASK) + >> CLOCK_RATE_SHIFT; + if (rate_index >= ARRAY_SIZE(dice_rates)) { err = -ENXIO; goto err_lock; } - rate = dice_rates[rate]; + rate = dice_rates[rate_index]; err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, rx_address(dice, RX_NUMBER_AUDIO), @@ -413,9 +414,20 @@ static int dice_open(struct snd_pcm_substream *substream) runtime->hw.channels_min = be32_to_cpu(number_audio); runtime->hw.channels_max = be32_to_cpu(number_audio); - amdtp_out_stream_set_rate(&dice->stream, rate); - amdtp_out_stream_set_pcm(&dice->stream, be32_to_cpu(number_audio)); - amdtp_out_stream_set_midi(&dice->stream, be32_to_cpu(number_midi)); + amdtp_out_stream_set_parameters(&dice->stream, rate, + be32_to_cpu(number_audio), + be32_to_cpu(number_midi)); + + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + amdtp_syt_intervals[rate_index]); + if (err < 0) + goto err_lock; + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + amdtp_syt_intervals[rate_index]); + if (err < 0) + goto err_lock; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, @@ -993,7 +1005,8 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) goto err_notification_handler; dice->resources.channels_mask = 0x00000000ffffffffuLL; - err = amdtp_out_stream_init(&dice->stream, unit, CIP_BLOCKING); + err = amdtp_out_stream_init(&dice->stream, unit, + CIP_BLOCKING | CIP_HI_DUALWIRE); if (err < 0) goto err_resources; diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 0ac56304baff..6a68caf1d04f 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -245,8 +245,10 @@ static int fwspk_hw_params(struct snd_pcm_substream *substream, if (err < 0) goto error; - amdtp_out_stream_set_rate(&fwspk->stream, params_rate(hw_params)); - amdtp_out_stream_set_pcm(&fwspk->stream, params_channels(hw_params)); + amdtp_out_stream_set_parameters(&fwspk->stream, + params_rate(hw_params), + params_channels(hw_params), + 0); amdtp_out_stream_set_pcm_format(&fwspk->stream, params_format(hw_params)); -- cgit From a644a9473f7f9519e2fe519136959dd0e671572a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:17:31 +0200 Subject: ALSA: dice: optimize reading of consecutive registers Instead of reading two consecutive register with two quadlet requests, use one block read request. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 27 +++++++++++---------------- 1 file changed, 11 insertions(+), 16 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 8804e42a96c6..e1d8dff23397 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -374,8 +374,8 @@ static int dice_open(struct snd_pcm_substream *substream) }; struct dice *dice = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - __be32 clock_sel, number_audio, number_midi; - unsigned int rate_index, rate; + __be32 clock_sel, data[2]; + unsigned int rate_index, number_audio, number_midi; int err; err = dice_try_lock(dice); @@ -393,30 +393,25 @@ static int dice_open(struct snd_pcm_substream *substream) err = -ENXIO; goto err_lock; } - rate = dice_rates[rate_index]; - err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, rx_address(dice, RX_NUMBER_AUDIO), - &number_audio, 4); - if (err < 0) - goto err_lock; - err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, - rx_address(dice, RX_NUMBER_MIDI), - &number_midi, 4); + data, 2 * 4); if (err < 0) goto err_lock; + number_audio = be32_to_cpu(data[0]); + number_midi = be32_to_cpu(data[1]); runtime->hw = hardware; - runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate); + runtime->hw.rates = snd_pcm_rate_to_rate_bit(dice_rates[rate_index]); snd_pcm_limit_hw_rates(runtime); - runtime->hw.channels_min = be32_to_cpu(number_audio); - runtime->hw.channels_max = be32_to_cpu(number_audio); + runtime->hw.channels_min = number_audio; + runtime->hw.channels_max = number_audio; - amdtp_out_stream_set_parameters(&dice->stream, rate, - be32_to_cpu(number_audio), - be32_to_cpu(number_midi)); + amdtp_out_stream_set_parameters(&dice->stream, dice_rates[rate_index], + number_audio, number_midi); err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -- cgit From 1b70485f135a39d5f2d8c392a16817456fa3a5cd Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:17:38 +0200 Subject: ALSA: firewire: extend snd_fw_transaction() Add a flag to snd_fw_transaction() to allow it to abort when a bus reset happens. This removes most of the duplicated error handling loops that were required around calls to the low-level fw_run_transaction(). Also add a flag to suppress error messages; errors are expected when we attempt to clean up after the device was unplugged. Signed-off-by: Clemens Ladisch --- sound/firewire/cmp.c | 50 +++++------ sound/firewire/dice.c | 207 ++++++++++++++-------------------------------- sound/firewire/fcp.c | 2 +- sound/firewire/isight.c | 43 +++++----- sound/firewire/lib.c | 24 ++++-- sound/firewire/lib.h | 7 +- sound/firewire/scs1x.c | 8 +- sound/firewire/speakers.c | 2 +- 8 files changed, 137 insertions(+), 206 deletions(-) diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index 645cb0ba4293..efdbf585e404 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -48,9 +48,6 @@ static int pcr_modify(struct cmp_connection *c, int (*check)(struct cmp_connection *c, __be32 pcr), enum bus_reset_handling bus_reset_handling) { - struct fw_device *device = fw_parent_device(c->resources.unit); - int generation = c->resources.generation; - int rcode, errors = 0; __be32 old_arg, buffer[2]; int err; @@ -59,36 +56,31 @@ static int pcr_modify(struct cmp_connection *c, old_arg = buffer[0]; buffer[1] = modify(c, buffer[0]); - rcode = fw_run_transaction( - device->card, TCODE_LOCK_COMPARE_SWAP, - device->node_id, generation, device->max_speed, + err = snd_fw_transaction( + c->resources.unit, TCODE_LOCK_COMPARE_SWAP, CSR_REGISTER_BASE + CSR_IPCR(c->pcr_index), - buffer, 8); - - if (rcode == RCODE_COMPLETE) { - if (buffer[0] == old_arg) /* success? */ - break; - - if (check) { - err = check(c, buffer[0]); - if (err < 0) - return err; - } - } else if (rcode == RCODE_GENERATION) - goto bus_reset; - else if (rcode_is_permanent_error(rcode) || ++errors >= 3) - goto io_error; + buffer, 8, + FW_FIXED_GENERATION | c->resources.generation); + + if (err < 0) { + if (err == -EAGAIN && + bus_reset_handling == SUCCEED_ON_BUS_RESET) + err = 0; + return err; + } + + if (buffer[0] == old_arg) /* success? */ + break; + + if (check) { + err = check(c, buffer[0]); + if (err < 0) + return err; + } } c->last_pcr_value = buffer[1]; return 0; - -io_error: - cmp_error(c, "transaction failed: %s\n", fw_rcode_string(rcode)); - return -EIO; - -bus_reset: - return bus_reset_handling == ABORT_ON_BUS_RESET ? -EAGAIN : 0; } @@ -108,7 +100,7 @@ int cmp_connection_init(struct cmp_connection *c, err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, CSR_REGISTER_BASE + CSR_IMPR, - &impr_be, 4); + &impr_be, 4, 0); if (err < 0) return err; impr = be32_to_cpu(impr_be); diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index e1d8dff23397..59d5ca4438b2 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -118,7 +118,7 @@ static int dice_owner_set(struct dice *dice) { struct fw_device *device = fw_parent_device(dice->unit); __be64 *buffer; - int rcode, err, errors = 0; + int err, errors = 0; buffer = kmalloc(2 * 8, GFP_KERNEL); if (!buffer) @@ -132,31 +132,24 @@ static int dice_owner_set(struct dice *dice) dice->owner_generation = device->generation; smp_rmb(); /* node_id vs. generation */ - rcode = fw_run_transaction(device->card, - TCODE_LOCK_COMPARE_SWAP, - device->node_id, - dice->owner_generation, - device->max_speed, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8); - - if (rcode == RCODE_COMPLETE) { - if (buffer[0] == cpu_to_be64(OWNER_NO_OWNER)) { - err = 0; - } else { + err = snd_fw_transaction(dice->unit, + TCODE_LOCK_COMPARE_SWAP, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8, + FW_FIXED_GENERATION | + dice->owner_generation); + + if (err == 0) { + if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { dev_err(&dice->unit->device, "device is already in use\n"); err = -EBUSY; } break; } - if (rcode_is_permanent_error(rcode) || ++errors >= 3) { - dev_err(&dice->unit->device, - "setting device owner failed: %s\n", - fw_rcode_string(rcode)); - err = -EIO; + if (err != -EAGAIN || ++errors >= 3) break; - } + msleep(20); } @@ -169,7 +162,7 @@ static int dice_owner_update(struct dice *dice) { struct fw_device *device = fw_parent_device(dice->unit); __be64 *buffer; - int rcode, err, errors = 0; + int err; if (dice->owner_generation == -1) return 0; @@ -178,44 +171,26 @@ static int dice_owner_update(struct dice *dice) if (!buffer) return -ENOMEM; - for (;;) { - buffer[0] = cpu_to_be64(OWNER_NO_OWNER); - buffer[1] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); + buffer[0] = cpu_to_be64(OWNER_NO_OWNER); + buffer[1] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); - dice->owner_generation = device->generation; - smp_rmb(); /* node_id vs. generation */ - rcode = fw_run_transaction(device->card, - TCODE_LOCK_COMPARE_SWAP, - device->node_id, - dice->owner_generation, - device->max_speed, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8); - - if (rcode == RCODE_COMPLETE) { - if (buffer[0] == cpu_to_be64(OWNER_NO_OWNER)) { - err = 0; - } else { - dev_err(&dice->unit->device, - "device is already in use\n"); - err = -EBUSY; - } - break; - } - if (rcode == RCODE_GENERATION) { - err = 0; /* try again later */ - break; - } - if (rcode_is_permanent_error(rcode) || ++errors >= 3) { + dice->owner_generation = device->generation; + smp_rmb(); /* node_id vs. generation */ + err = snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8, + FW_FIXED_GENERATION | dice->owner_generation); + + if (err == 0) { + if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { dev_err(&dice->unit->device, - "setting device owner failed: %s\n", - fw_rcode_string(rcode)); - err = -EIO; - break; + "device is already in use\n"); + err = -EBUSY; } - msleep(20); + } else if (err == -EAGAIN) { + err = 0; /* try again later */ } kfree(buffer); @@ -230,38 +205,19 @@ static void dice_owner_clear(struct dice *dice) { struct fw_device *device = fw_parent_device(dice->unit); __be64 *buffer; - int rcode, errors = 0; buffer = kmalloc(2 * 8, GFP_KERNEL); if (!buffer) return; - for (;;) { - buffer[0] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); - buffer[1] = cpu_to_be64(OWNER_NO_OWNER); - - rcode = fw_run_transaction(device->card, - TCODE_LOCK_COMPARE_SWAP, - device->node_id, - dice->owner_generation, - device->max_speed, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8); - - if (rcode == RCODE_COMPLETE) - break; - if (rcode == RCODE_GENERATION) - break; - if (rcode_is_permanent_error(rcode) || ++errors >= 3) { - dev_err(&dice->unit->device, - "clearing device owner failed: %s\n", - fw_rcode_string(rcode)); - break; - } - msleep(20); - } + buffer[0] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + buffer[1] = cpu_to_be64(OWNER_NO_OWNER); + snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8, FW_QUIET | + FW_FIXED_GENERATION | dice->owner_generation); kfree(buffer); @@ -270,67 +226,32 @@ static void dice_owner_clear(struct dice *dice) static int dice_enable_set(struct dice *dice) { - struct fw_device *device = fw_parent_device(dice->unit); __be32 value; - int rcode, err, errors = 0; + int err; value = cpu_to_be32(1); - for (;;) { - rcode = fw_run_transaction(device->card, - TCODE_WRITE_QUADLET_REQUEST, - device->node_id, - dice->owner_generation, - device->max_speed, - global_address(dice, GLOBAL_ENABLE), - &value, 4); - if (rcode == RCODE_COMPLETE) { - dice->global_enabled = true; - err = 0; - break; - } - if (rcode == RCODE_GENERATION) { - err = -EAGAIN; - break; - } - if (rcode_is_permanent_error(rcode) || ++errors >= 3) { - dev_err(&dice->unit->device, - "device enabling failed: %s\n", - fw_rcode_string(rcode)); - err = -EIO; - break; - } - msleep(20); - } + err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_ENABLE), + &value, 4, + FW_FIXED_GENERATION | dice->owner_generation); + if (err < 0) + return err; - return err; + dice->global_enabled = true; + + return 0; } static void dice_enable_clear(struct dice *dice) { - struct fw_device *device = fw_parent_device(dice->unit); __be32 value; - int rcode, errors = 0; value = 0; - for (;;) { - rcode = fw_run_transaction(device->card, - TCODE_WRITE_QUADLET_REQUEST, - device->node_id, - dice->owner_generation, - device->max_speed, - global_address(dice, GLOBAL_ENABLE), - &value, 4); - if (rcode == RCODE_COMPLETE || - rcode == RCODE_GENERATION) - break; - if (rcode_is_permanent_error(rcode) || ++errors >= 3) { - dev_err(&dice->unit->device, - "device disabling failed: %s\n", - fw_rcode_string(rcode)); - break; - } - msleep(20); - } + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_ENABLE), + &value, 4, FW_QUIET | + FW_FIXED_GENERATION | dice->owner_generation); + dice->global_enabled = false; } @@ -384,7 +305,7 @@ static int dice_open(struct snd_pcm_substream *substream) err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4); + &clock_sel, 4, 0); if (err < 0) goto err_lock; rate_index = (be32_to_cpu(clock_sel) & CLOCK_RATE_MASK) @@ -396,7 +317,7 @@ static int dice_open(struct snd_pcm_substream *substream) err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, rx_address(dice, RX_NUMBER_AUDIO), - data, 2 * 4); + data, 2 * 4, 0); if (err < 0) goto err_lock; number_audio = be32_to_cpu(data[0]); @@ -488,7 +409,7 @@ static int dice_stream_start(struct dice *dice) err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, rx_address(dice, RX_ISOCHRONOUS), - &channel, 4); + &channel, 4, 0); if (err < 0) goto err_resources; } @@ -502,7 +423,7 @@ static int dice_stream_start(struct dice *dice) err_rx_channel: channel = cpu_to_be32((u32)-1); snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), &channel, 4); + rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); err_resources: fw_iso_resources_free(&dice->resources); error: @@ -528,7 +449,7 @@ static void dice_stream_stop(struct dice *dice) channel = cpu_to_be32((u32)-1); snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), &channel, 4); + rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); fw_iso_resources_free(&dice->resources); } @@ -880,7 +801,7 @@ static int dice_interface_check(struct fw_unit *unit) */ err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, DICE_PRIVATE_SPACE, - pointers, sizeof(pointers)); + pointers, sizeof(pointers), 0); if (err < 0) return -ENODEV; for (i = 0; i < ARRAY_SIZE(pointers); ++i) { @@ -896,7 +817,7 @@ static int dice_interface_check(struct fw_unit *unit) err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, DICE_PRIVATE_SPACE + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, - &version, 4); + &version, 4, 0); if (err < 0) return -ENODEV; if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { @@ -915,7 +836,7 @@ static int dice_init_offsets(struct dice *dice) err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, DICE_PRIVATE_SPACE, - pointers, sizeof(pointers)); + pointers, sizeof(pointers), 0); if (err < 0) return err; @@ -939,7 +860,7 @@ static void dice_card_strings(struct dice *dice) BUILD_BUG_ON(NICK_NAME_SIZE < sizeof(card->shortname)); err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, global_address(dice, GLOBAL_NICK_NAME), - card->shortname, sizeof(card->shortname)); + card->shortname, sizeof(card->shortname), 0); if (err >= 0) { /* DICE strings are returned in "always-wrong" endianness */ BUILD_BUG_ON(sizeof(card->shortname) % 4 != 0); @@ -1015,14 +936,14 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4); + &clock_sel, 4, 0); if (err < 0) goto error; clock_sel &= cpu_to_be32(~CLOCK_SOURCE_MASK); clock_sel |= cpu_to_be32(CLOCK_SOURCE_ARX1); err = snd_fw_transaction(unit, TCODE_WRITE_QUADLET_REQUEST, global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4); + &clock_sel, 4, 0); if (err < 0) goto error; diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index ec578b5ad8da..860c08073c59 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -90,7 +90,7 @@ int fcp_avc_transaction(struct fw_unit *unit, : TCODE_WRITE_BLOCK_REQUEST; ret = snd_fw_transaction(t.unit, tcode, CSR_REGISTER_BASE + CSR_FCP_COMMAND, - (void *)command, command_size); + (void *)command, command_size, 0); if (ret < 0) break; diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 58a5afefdc69..fd42e6b315e6 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -217,7 +217,7 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle, static int isight_connect(struct isight *isight) { - int ch, err, rcode, errors = 0; + int ch, err; __be32 value; retry_after_bus_reset: @@ -230,27 +230,19 @@ retry_after_bus_reset: } value = cpu_to_be32(ch | (isight->device->max_speed << SPEED_SHIFT)); - for (;;) { - rcode = fw_run_transaction( - isight->device->card, - TCODE_WRITE_QUADLET_REQUEST, - isight->device->node_id, - isight->resources.generation, - isight->device->max_speed, - isight->audio_base + REG_ISO_TX_CONFIG, - &value, 4); - if (rcode == RCODE_COMPLETE) { - return 0; - } else if (rcode == RCODE_GENERATION) { - fw_iso_resources_free(&isight->resources); - goto retry_after_bus_reset; - } else if (rcode_is_permanent_error(rcode) || ++errors >= 3) { - err = -EIO; - goto err_resources; - } - msleep(5); + err = snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_ISO_TX_CONFIG, + &value, 4, FW_FIXED_GENERATION | + isight->resources.generation); + if (err == -EAGAIN) { + fw_iso_resources_free(&isight->resources); + goto retry_after_bus_reset; + } else if (err < 0) { + goto err_resources; } + return 0; + err_resources: fw_iso_resources_free(&isight->resources); error: @@ -315,17 +307,19 @@ static int isight_hw_params(struct snd_pcm_substream *substream, static int reg_read(struct isight *isight, int offset, __be32 *value) { return snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, - isight->audio_base + offset, value, 4); + isight->audio_base + offset, value, 4, 0); } static int reg_write(struct isight *isight, int offset, __be32 value) { return snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, - isight->audio_base + offset, &value, 4); + isight->audio_base + offset, &value, 4, 0); } static void isight_stop_streaming(struct isight *isight) { + __be32 value; + if (!isight->context) return; @@ -333,7 +327,10 @@ static void isight_stop_streaming(struct isight *isight) fw_iso_context_destroy(isight->context); isight->context = NULL; fw_iso_resources_free(&isight->resources); - reg_write(isight, REG_AUDIO_ENABLE, 0); + value = 0; + snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_AUDIO_ENABLE, + &value, 4, FW_QUIET); } static int isight_hw_free(struct snd_pcm_substream *substream) diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index 14eb41498372..7409edba9f06 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -11,7 +11,7 @@ #include #include "lib.h" -#define ERROR_RETRY_DELAY_MS 5 +#define ERROR_RETRY_DELAY_MS 20 /** * snd_fw_transaction - send a request and wait for its completion @@ -20,6 +20,9 @@ * @offset: the address in the target's address space * @buffer: input/output data * @length: length of @buffer + * @flags: use %FW_FIXED_GENERATION and add the generation value to attempt the + * request only in that generation; use %FW_QUIET to suppress error + * messages * * Submits an asynchronous request to the target device, and waits for the * response. The node ID and the current generation are derived from @unit. @@ -27,14 +30,18 @@ * Returns zero on success, or a negative error code. */ int snd_fw_transaction(struct fw_unit *unit, int tcode, - u64 offset, void *buffer, size_t length) + u64 offset, void *buffer, size_t length, + unsigned int flags) { struct fw_device *device = fw_parent_device(unit); int generation, rcode, tries = 0; + generation = flags & FW_GENERATION_MASK; for (;;) { - generation = device->generation; - smp_rmb(); /* node_id vs. generation */ + if (!(flags & FW_FIXED_GENERATION)) { + generation = device->generation; + smp_rmb(); /* node_id vs. generation */ + } rcode = fw_run_transaction(device->card, tcode, device->node_id, generation, device->max_speed, offset, @@ -43,9 +50,14 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode, if (rcode == RCODE_COMPLETE) return 0; + if (rcode == RCODE_GENERATION && (flags & FW_FIXED_GENERATION)) + return -EAGAIN; + if (rcode_is_permanent_error(rcode) || ++tries >= 3) { - dev_err(&unit->device, "transaction failed: %s\n", - fw_rcode_string(rcode)); + if (!(flags & FW_QUIET)) + dev_err(&unit->device, + "transaction failed: %s\n", + fw_rcode_string(rcode)); return -EIO; } diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h index aef301476ea9..02cfabc9c3c4 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -6,8 +6,13 @@ struct fw_unit; +#define FW_GENERATION_MASK 0x00ff +#define FW_FIXED_GENERATION 0x0100 +#define FW_QUIET 0x0200 + int snd_fw_transaction(struct fw_unit *unit, int tcode, - u64 offset, void *buffer, size_t length); + u64 offset, void *buffer, size_t length, + unsigned int flags); /* returns true if retrying the transaction would not make sense */ static inline bool rcode_is_permanent_error(int rcode) diff --git a/sound/firewire/scs1x.c b/sound/firewire/scs1x.c index 505fc8123199..858023cf4298 100644 --- a/sound/firewire/scs1x.c +++ b/sound/firewire/scs1x.c @@ -369,7 +369,7 @@ static int scs_init_hss_address(struct scs *scs) data = cpu_to_be64(((u64)HSS1394_TAG_CHANGE_ADDRESS << 56) | scs->hss_handler.offset); err = snd_fw_transaction(scs->unit, TCODE_WRITE_BLOCK_REQUEST, - HSS1394_ADDRESS, &data, 8); + HSS1394_ADDRESS, &data, 8, 0); if (err < 0) dev_err(&scs->unit->device, "HSS1394 communication failed\n"); @@ -455,12 +455,16 @@ err_card: static void scs_update(struct fw_unit *unit) { struct scs *scs = dev_get_drvdata(&unit->device); + int generation; __be64 data; data = cpu_to_be64(((u64)HSS1394_TAG_CHANGE_ADDRESS << 56) | scs->hss_handler.offset); + generation = fw_parent_device(unit)->generation; + smp_rmb(); /* node_id vs. generation */ snd_fw_transaction(scs->unit, TCODE_WRITE_BLOCK_REQUEST, - HSS1394_ADDRESS, &data, 8); + HSS1394_ADDRESS, &data, 8, + FW_FIXED_GENERATION | generation); } static void scs_remove(struct fw_unit *unit) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 6a68caf1d04f..eb3f7dce1d6c 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -647,7 +647,7 @@ static u32 fwspk_read_firmware_version(struct fw_unit *unit) int err; err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, - OXFORD_FIRMWARE_ID_ADDRESS, &data, 4); + OXFORD_FIRMWARE_ID_ADDRESS, &data, 4, 0); return err >= 0 ? be32_to_cpu(data) : 0; } -- cgit From eadce07faa8e71d8a0fc7501a5167fb999200225 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:17:45 +0200 Subject: ALSA: dice: avoid superflous write at bus reset When a bus reset happens, the enable register is automatically cleared, so we do not need to clear it manually when stopping the stream. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 59d5ca4438b2..cfa98a83acb2 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -246,6 +246,9 @@ static void dice_enable_clear(struct dice *dice) { __be32 value; + if (!dice->global_enabled) + return; + value = 0; snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, global_address(dice, GLOBAL_ENABLE), @@ -1009,6 +1012,8 @@ static void dice_bus_reset(struct fw_unit *unit) * manner. */ amdtp_out_stream_pcm_abort(&dice->stream); + + dice->global_enabled = false; dice_stream_stop_packets(dice); dice_owner_update(dice); -- cgit From 435a9be8bdb7422b82c194fad2d279ef436addc1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:17:51 +0200 Subject: ALSA: dice: remove 10s period length limit Since commit f2b3614cefb6 (Don't check DMA time-out too shortly), we need no longer to restrict the period length to less than 10 s. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index cfa98a83acb2..57ceb13f2815 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -350,7 +350,7 @@ static int dice_open(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 5000, 8192000); + 5000, UINT_MAX); if (err < 0) goto err_lock; -- cgit From 8709f1e4d68b0b3caf9783cf2463e5747943bff8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 11 Oct 2011 17:51:16 +0200 Subject: ALSA: dice: remove superfluous field The pcm field was not actually used. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 57ceb13f2815..2d198aed6f94 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -43,7 +43,6 @@ struct dice { bool global_enabled; wait_queue_head_t hwdep_wait; u32 notification_bits; - struct snd_pcm_substream *pcm; struct fw_iso_resources resources; struct amdtp_out_stream stream; }; @@ -564,8 +563,7 @@ static int dice_create_pcm(struct dice *dice) return err; pcm->private_data = dice; strcpy(pcm->name, dice->card->shortname); - dice->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - dice->pcm->ops = &ops; + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->ops = &ops; return 0; } -- cgit From a8c558f6a3eedfb9bfd7d9d82f9d00f2f807ce7c Mon Sep 17 00:00:00 2001 From: Stefan Richter Date: Sat, 27 Aug 2011 20:05:15 +0200 Subject: ALSA: dice: fix locking Avoid a lock inversion between dice->mutex and pcm->open_mutex. Signed-off-by: Stefan Richter Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 2d198aed6f94..2d3a04ef9840 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -981,12 +981,12 @@ static void dice_remove(struct fw_unit *unit) { struct dice *dice = dev_get_drvdata(&unit->device); - mutex_lock(&dice->mutex); - amdtp_out_stream_pcm_abort(&dice->stream); snd_card_disconnect(dice->card); + mutex_lock(&dice->mutex); + dice_stream_stop(dice); dice_owner_clear(dice); @@ -999,8 +999,6 @@ static void dice_bus_reset(struct fw_unit *unit) { struct dice *dice = dev_get_drvdata(&unit->device); - mutex_lock(&dice->mutex); - /* * On a bus reset, the DICE firmware disables streaming and then goes * off contemplating its own navel for hundreds of milliseconds before @@ -1011,6 +1009,8 @@ static void dice_bus_reset(struct fw_unit *unit) */ amdtp_out_stream_pcm_abort(&dice->stream); + mutex_lock(&dice->mutex); + dice->global_enabled = false; dice_stream_stop_packets(dice); -- cgit From c5280e996fdd14ffacfbbdf8ce5ac7f913b101d2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 16 Oct 2011 21:39:00 +0200 Subject: ALSA: dice: make amdtp_rates[] const Signed-off-by: Clemens Ladisch --- sound/firewire/amdtp.c | 2 +- sound/firewire/amdtp.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index a09c3b393c0b..5540f707bfdb 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -65,7 +65,7 @@ void amdtp_out_stream_destroy(struct amdtp_out_stream *s) } EXPORT_SYMBOL(amdtp_out_stream_destroy); -unsigned int amdtp_syt_intervals[CIP_SFC_COUNT] = { +const unsigned int amdtp_syt_intervals[CIP_SFC_COUNT] = { [CIP_SFC_32000] = 8, [CIP_SFC_44100] = 8, [CIP_SFC_48000] = 8, diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index f3d03dd92c39..839ebf812d79 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -103,7 +103,7 @@ void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s); unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s); void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s); -extern unsigned int amdtp_syt_intervals[CIP_SFC_COUNT]; +extern const unsigned int amdtp_syt_intervals[CIP_SFC_COUNT]; static inline bool amdtp_out_stream_running(struct amdtp_out_stream *s) { -- cgit From a0301998aeebad206fa7e1b77300c9961c8c3d12 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Dec 2011 21:47:00 +0100 Subject: ALSA: dice: get clock capabilities In preparation for sample rate selection support, ensure that the driver knows about the device's clock capabilities. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 2d3a04ef9840..06fef4752d50 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -36,6 +36,7 @@ struct dice { struct mutex mutex; unsigned int global_offset; unsigned int rx_offset; + unsigned int clock_caps; struct fw_address_handler notification_handler; int owner_generation; int dev_lock_count; /* > 0 driver, < 0 userspace */ @@ -830,9 +831,10 @@ static int dice_interface_check(struct fw_unit *unit) return 0; } -static int dice_init_offsets(struct dice *dice) +static int dice_read_params(struct dice *dice) { __be32 pointers[6]; + __be32 value; int err; err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, @@ -844,6 +846,23 @@ static int dice_init_offsets(struct dice *dice) dice->global_offset = be32_to_cpu(pointers[0]) * 4; dice->rx_offset = be32_to_cpu(pointers[4]) * 4; + /* some very old firmwares don't tell about their clock support */ + if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) { + err = snd_fw_transaction( + dice->unit, TCODE_READ_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_CAPABILITIES), + &value, 4, 0); + if (err < 0) + return err; + dice->clock_caps = be32_to_cpu(value); + } else { + /* this should be supported by any device */ + dice->clock_caps = CLOCK_CAP_RATE_44100 | + CLOCK_CAP_RATE_48000 | + CLOCK_CAP_SOURCE_ARX1 | + CLOCK_CAP_SOURCE_INTERNAL; + } + return 0; } @@ -905,7 +924,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice->unit = unit; init_waitqueue_head(&dice->hwdep_wait); - err = dice_init_offsets(dice); + err = dice_read_params(dice); if (err < 0) goto err_mutex; -- cgit From 5ea4018e4321f24e8305ea8a8b0a9c3e270456ae Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 5 Dec 2011 22:09:42 +0100 Subject: ALSA: dice: allow notifications during initialization Reorganize the initialization order so that the driver can receive notifications earlier. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 06fef4752d50..49b47ba09cdd 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -924,10 +924,6 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice->unit = unit; init_waitqueue_head(&dice->hwdep_wait); - err = dice_read_params(dice); - if (err < 0) - goto err_mutex; - dice->notification_handler.length = 4; dice->notification_handler.address_callback = dice_notification; dice->notification_handler.callback_data = dice; @@ -936,9 +932,17 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) if (err < 0) goto err_mutex; - err = fw_iso_resources_init(&dice->resources, unit); + err = dice_owner_set(dice); if (err < 0) goto err_notification_handler; + + err = dice_read_params(dice); + if (err < 0) + goto err_owner; + + err = fw_iso_resources_init(&dice->resources, unit); + if (err < 0) + goto err_owner; dice->resources.channels_mask = 0x00000000ffffffffuLL; err = amdtp_out_stream_init(&dice->stream, unit, @@ -946,10 +950,6 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) if (err < 0) goto err_resources; - err = dice_owner_set(dice); - if (err < 0) - goto err_stream; - card->private_free = dice_card_free; dice_card_strings(dice); @@ -983,10 +983,10 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) return 0; -err_stream: - amdtp_out_stream_destroy(&dice->stream); err_resources: fw_iso_resources_destroy(&dice->resources); +err_owner: + dice_owner_clear(dice); err_notification_handler: fw_core_remove_address_handler(&dice->notification_handler); err_mutex: -- cgit From 15a75c8bed591dd23a3d221f5ccd91843c109670 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Dec 2011 22:23:59 +0100 Subject: ALSA: dice: get rate-dependent parameters In preparation for sample rate selection support, read the stream parameters that might change when running at different sample rates. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 92 +++++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 90 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 49b47ba09cdd..e6bba6d32bd8 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -6,10 +6,12 @@ */ #include +#include #include #include #include #include +#include #include #include #include @@ -37,11 +39,14 @@ struct dice { unsigned int global_offset; unsigned int rx_offset; unsigned int clock_caps; + unsigned int rx_channels[3]; + unsigned int rx_midi_ports[3]; struct fw_address_handler notification_handler; int owner_generation; int dev_lock_count; /* > 0 driver, < 0 userspace */ bool dev_lock_changed; bool global_enabled; + struct completion clock_accepted; wait_queue_head_t hwdep_wait; u32 notification_bits; struct fw_iso_resources resources; @@ -53,15 +58,23 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); static const unsigned int dice_rates[] = { + /* mode 0 */ [0] = 32000, [1] = 44100, [2] = 48000, + /* mode 1 */ [3] = 88200, [4] = 96000, + /* mode 2 */ [5] = 176400, [6] = 192000, }; +static unsigned int rate_index_to_mode(unsigned int rate_index) +{ + return ((int)rate_index - 1) / 2; +} + static void dice_lock_changed(struct dice *dice) { dice->dev_lock_changed = true; @@ -264,6 +277,7 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, void *data, size_t length, void *callback_data) { struct dice *dice = callback_data; + u32 bits; unsigned long flags; if (tcode != TCODE_WRITE_QUADLET_REQUEST) { @@ -274,10 +288,17 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, fw_send_response(card, request, RCODE_ADDRESS_ERROR); return; } + + bits = be32_to_cpup(data); + spin_lock_irqsave(&dice->lock, flags); - dice->notification_bits |= be32_to_cpup(data); + dice->notification_bits |= bits; spin_unlock_irqrestore(&dice->lock, flags); + fw_send_response(card, request, RCODE_COMPLETE); + + if (bits & NOTIFY_CLOCK_ACCEPTED) + complete(&dice->clock_accepted); wake_up(&dice->hwdep_wait); } @@ -457,6 +478,26 @@ static void dice_stream_stop(struct dice *dice) fw_iso_resources_free(&dice->resources); } +static int dice_change_rate(struct dice *dice, unsigned int clock_rate) +{ + __be32 value; + int err; + + INIT_COMPLETION(dice->clock_accepted); + + value = cpu_to_be32(clock_rate | CLOCK_SOURCE_ARX1); + err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &value, 4, 0); + if (err < 0) + return err; + + wait_for_completion_timeout(&dice->clock_accepted, + msecs_to_jiffies(100)); + + return 0; +} + static int dice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -831,11 +872,51 @@ static int dice_interface_check(struct fw_unit *unit) return 0; } +static int highest_supported_mode_rate(struct dice *dice, unsigned int mode) +{ + int i; + + for (i = ARRAY_SIZE(dice_rates) - 1; i >= 0; --i) + if ((dice->clock_caps & (1 << i)) && + rate_index_to_mode(i) == mode) + return i; + + return -1; +} + +static int dice_read_mode_params(struct dice *dice, unsigned int mode) +{ + __be32 values[2]; + int rate_index, err; + + rate_index = highest_supported_mode_rate(dice, mode); + if (rate_index < 0) { + dice->rx_channels[mode] = 0; + dice->rx_midi_ports[mode] = 0; + return 0; + } + + err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); + if (err < 0) + return err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + rx_address(dice, RX_NUMBER_AUDIO), + values, 2 * 4, 0); + if (err < 0) + return err; + + dice->rx_channels[mode] = be32_to_cpu(values[0]); + dice->rx_midi_ports[mode] = be32_to_cpu(values[1]); + + return 0; +} + static int dice_read_params(struct dice *dice) { __be32 pointers[6]; __be32 value; - int err; + int mode, err; err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, DICE_PRIVATE_SPACE, @@ -863,6 +944,12 @@ static int dice_read_params(struct dice *dice) CLOCK_CAP_SOURCE_INTERNAL; } + for (mode = 2; mode >= 0; --mode) { + err = dice_read_mode_params(dice, mode); + if (err < 0) + return err; + } + return 0; } @@ -922,6 +1009,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) spin_lock_init(&dice->lock); mutex_init(&dice->mutex); dice->unit = unit; + init_completion(&dice->clock_accepted); init_waitqueue_head(&dice->hwdep_wait); dice->notification_handler.length = 4; -- cgit From 4edeb831f32d17fba056eb752f7afc26a19674a0 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Dec 2011 22:07:01 +0100 Subject: ALSA: dice: dynamic sample rate selection Instead of relying of some control panel application to configure some fixed sample rate, allow applications to set it automatically. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 137 +++++++++++++++++++++++++++++++++++++------------- 1 file changed, 102 insertions(+), 35 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index e6bba6d32bd8..61dd00c4fae3 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -70,6 +70,17 @@ static const unsigned int dice_rates[] = { [6] = 192000, }; +static unsigned int rate_to_index(unsigned int rate) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + if (dice_rates[i] == rate) + return i; + + return 0; +} + static unsigned int rate_index_to_mode(unsigned int rate_index) { return ((int)rate_index - 1) / 2; @@ -302,6 +313,59 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, wake_up(&dice->hwdep_wait); } +static int dice_rate_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct dice *dice = rule->private; + const struct snd_interval *channels = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval allowed_rates = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int i, mode; + + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) { + mode = rate_index_to_mode(i); + if ((dice->clock_caps & (1 << i)) && + snd_interval_test(channels, dice->rx_channels[mode])) { + allowed_rates.min = min(allowed_rates.min, + dice_rates[i]); + allowed_rates.max = max(allowed_rates.max, + dice_rates[i]); + } + } + + return snd_interval_refine(rate, &allowed_rates); +} + +static int dice_channels_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct dice *dice = rule->private; + const struct snd_interval *rate = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval allowed_channels = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int i, mode; + + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + if ((dice->clock_caps & (1 << i)) && + snd_interval_test(rate, dice_rates[i])) { + mode = rate_index_to_mode(i); + allowed_channels.min = min(allowed_channels.min, + dice->rx_channels[mode]); + allowed_channels.max = max(allowed_channels.max, + dice->rx_channels[mode]); + } + + return snd_interval_refine(channels, &allowed_channels); +} + static int dice_open(struct snd_pcm_substream *substream) { static const struct snd_pcm_hardware hardware = { @@ -311,6 +375,8 @@ static int dice_open(struct snd_pcm_substream *substream) SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .formats = AMDTP_OUT_PCM_FORMAT_BITS, + .channels_min = UINT_MAX, + .channels_max = 0, .buffer_bytes_max = 16 * 1024 * 1024, .period_bytes_min = 1, .period_bytes_max = UINT_MAX, @@ -319,53 +385,46 @@ static int dice_open(struct snd_pcm_substream *substream) }; struct dice *dice = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - __be32 clock_sel, data[2]; - unsigned int rate_index, number_audio, number_midi; + unsigned int i; int err; err = dice_try_lock(dice); if (err < 0) goto error; - err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4, 0); - if (err < 0) - goto err_lock; - rate_index = (be32_to_cpu(clock_sel) & CLOCK_RATE_MASK) - >> CLOCK_RATE_SHIFT; - if (rate_index >= ARRAY_SIZE(dice_rates)) { - err = -ENXIO; - goto err_lock; - } - - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - rx_address(dice, RX_NUMBER_AUDIO), - data, 2 * 4, 0); - if (err < 0) - goto err_lock; - number_audio = be32_to_cpu(data[0]); - number_midi = be32_to_cpu(data[1]); - runtime->hw = hardware; - runtime->hw.rates = snd_pcm_rate_to_rate_bit(dice_rates[rate_index]); + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + if (dice->clock_caps & (1 << i)) + runtime->hw.rates |= + snd_pcm_rate_to_rate_bit(dice_rates[i]); snd_pcm_limit_hw_rates(runtime); - runtime->hw.channels_min = number_audio; - runtime->hw.channels_max = number_audio; + for (i = 0; i < 3; ++i) + if (dice->rx_channels[i]) { + runtime->hw.channels_min = min(runtime->hw.channels_min, + dice->rx_channels[i]); + runtime->hw.channels_max = max(runtime->hw.channels_max, + dice->rx_channels[i]); + } - amdtp_out_stream_set_parameters(&dice->stream, dice_rates[rate_index], - number_audio, number_midi); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + dice_rate_constraint, dice, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + goto err_lock; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + dice_channels_constraint, dice, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto err_lock; err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - amdtp_syt_intervals[rate_index]); + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); if (err < 0) goto err_lock; err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - amdtp_syt_intervals[rate_index]); + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); if (err < 0) goto err_lock; @@ -502,6 +561,7 @@ static int dice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct dice *dice = substream->private_data; + unsigned int rate_index, mode; int err; mutex_lock(&dice->mutex); @@ -511,15 +571,22 @@ static int dice_hw_params(struct snd_pcm_substream *substream, err = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (err < 0) - goto error; + return err; + rate_index = rate_to_index(params_rate(hw_params)); + err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); + if (err < 0) + return err; + + mode = rate_index_to_mode(rate_index); + amdtp_out_stream_set_parameters(&dice->stream, + params_rate(hw_params), + params_channels(hw_params), + dice->rx_midi_ports[mode]); amdtp_out_stream_set_pcm_format(&dice->stream, params_format(hw_params)); return 0; - -error: - return err; } static int dice_hw_free(struct snd_pcm_substream *substream) -- cgit From 640d9b421d4d8c7593aa8647479a4c7c6fe0ca65 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 5 Jan 2012 22:16:24 +0100 Subject: ALSA: dice: check clock change timeout Output a warning if the wait for the clock change notification times out. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 61dd00c4fae3..3395c8ba7af1 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -551,8 +551,9 @@ static int dice_change_rate(struct dice *dice, unsigned int clock_rate) if (err < 0) return err; - wait_for_completion_timeout(&dice->clock_accepted, - msecs_to_jiffies(100)); + if (!wait_for_completion_timeout(&dice->clock_accepted, + msecs_to_jiffies(100))) + dev_warn(&dice->unit->device, "clock change timed out\n"); return 0; } -- cgit From c614475b0ea9f7e6b3f76a46be315579bb899397 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 5 Jan 2012 22:36:08 +0100 Subject: ALSA: dice: add a proc file to show device information For easier debugging, add a proc file to show the device's capabilities and current status. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 246 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 246 insertions(+) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 3395c8ba7af1..25a96362e1aa 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include @@ -857,6 +858,249 @@ static int dice_create_hwdep(struct dice *dice) return 0; } +static int dice_proc_read_mem(struct dice *dice, void *buffer, + unsigned int offset_q, unsigned int quadlets) +{ + unsigned int i; + int err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE + 4 * offset_q, + buffer, 4 * quadlets, 0); + if (err < 0) + return err; + + for (i = 0; i < quadlets; ++i) + be32_to_cpus(&((u32 *)buffer)[i]); + + return 0; +} + +static const char *str_from_array(const char *const strs[], unsigned int count, + unsigned int i) +{ + if (i < count) + return strs[i]; + else + return "(unknown)"; +} + +static void dice_proc_fixup_string(char *s, unsigned int size) +{ + unsigned int i; + + for (i = 0; i < size; i += 4) + cpu_to_le32s((u32 *)(s + i)); + + for (i = 0; i < size - 2; ++i) { + if (s[i] == '\0') + return; + if (s[i] == '\\' && s[i + 1] == '\\') { + s[i + 2] = '\0'; + return; + } + } + s[size - 1] = '\0'; +} + +static void dice_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + static const char *const section_names[5] = { + "global", "tx", "rx", "ext_sync", "unused2" + }; + static const char *const clock_sources[] = { + "aes1", "aes2", "aes3", "aes4", "aes", "adat", "tdif", + "wc", "arx1", "arx2", "arx3", "arx4", "internal" + }; + static const char *const rates[] = { + "32000", "44100", "48000", "88200", "96000", "176400", "192000", + "any low", "any mid", "any high", "none" + }; + struct dice *dice = entry->private_data; + u32 sections[ARRAY_SIZE(section_names) * 2]; + struct { + u32 number; + u32 size; + } tx_rx_header; + union { + struct { + u32 owner_hi, owner_lo; + u32 notification; + char nick_name[NICK_NAME_SIZE]; + u32 clock_select; + u32 enable; + u32 status; + u32 extended_status; + u32 sample_rate; + u32 version; + u32 clock_caps; + char clock_source_names[CLOCK_SOURCE_NAMES_SIZE]; + } global; + struct { + u32 iso; + u32 number_audio; + u32 number_midi; + u32 speed; + char names[TX_NAMES_SIZE]; + u32 ac3_caps; + u32 ac3_enable; + } tx; + struct { + u32 iso; + u32 seq_start; + u32 number_audio; + u32 number_midi; + char names[RX_NAMES_SIZE]; + u32 ac3_caps; + u32 ac3_enable; + } rx; + struct { + u32 clock_source; + u32 locked; + u32 rate; + u32 adat_user_data; + } ext_sync; + } buf; + unsigned int quadlets, stream, i; + + if (dice_proc_read_mem(dice, sections, 0, ARRAY_SIZE(sections)) < 0) + return; + snd_iprintf(buffer, "sections:\n"); + for (i = 0; i < ARRAY_SIZE(section_names); ++i) + snd_iprintf(buffer, " %s: offset %u, size %u\n", + section_names[i], + sections[i * 2], sections[i * 2 + 1]); + + quadlets = min_t(u32, sections[1], sizeof(buf.global) / 4); + if (dice_proc_read_mem(dice, &buf.global, sections[0], quadlets) < 0) + return; + snd_iprintf(buffer, "global:\n"); + snd_iprintf(buffer, " owner: %04x:%04x%08x\n", + buf.global.owner_hi >> 16, + buf.global.owner_hi & 0xffff, buf.global.owner_lo); + snd_iprintf(buffer, " notification: %08x\n", buf.global.notification); + dice_proc_fixup_string(buf.global.nick_name, NICK_NAME_SIZE); + snd_iprintf(buffer, " nick name: %s\n", buf.global.nick_name); + snd_iprintf(buffer, " clock select: %s %s\n", + str_from_array(clock_sources, ARRAY_SIZE(clock_sources), + buf.global.clock_select & CLOCK_SOURCE_MASK), + str_from_array(rates, ARRAY_SIZE(rates), + (buf.global.clock_select & CLOCK_RATE_MASK) + >> CLOCK_RATE_SHIFT)); + snd_iprintf(buffer, " enable: %u\n", buf.global.enable); + snd_iprintf(buffer, " status: %slocked %s\n", + buf.global.status & STATUS_SOURCE_LOCKED ? "" : "un", + str_from_array(rates, ARRAY_SIZE(rates), + (buf.global.status & + STATUS_NOMINAL_RATE_MASK) + >> CLOCK_RATE_SHIFT)); + snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); + snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); + snd_iprintf(buffer, " version: %u.%u.%u.%u\n", + (buf.global.version >> 24) & 0xff, + (buf.global.version >> 16) & 0xff, + (buf.global.version >> 8) & 0xff, + (buf.global.version >> 0) & 0xff); + if (quadlets >= 90) { + snd_iprintf(buffer, " clock caps:"); + for (i = 0; i <= 6; ++i) + if (buf.global.clock_caps & (1 << i)) + snd_iprintf(buffer, " %s", rates[i]); + for (i = 0; i <= 12; ++i) + if (buf.global.clock_caps & (1 << (16 + i))) + snd_iprintf(buffer, " %s", clock_sources[i]); + snd_iprintf(buffer, "\n"); + dice_proc_fixup_string(buf.global.clock_source_names, + CLOCK_SOURCE_NAMES_SIZE); + snd_iprintf(buffer, " clock source names: %s\n", + buf.global.clock_source_names); + } + + if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) + return; + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx)); + for (stream = 0; stream < tx_rx_header.number; ++stream) { + if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + + stream * tx_rx_header.size, + quadlets) < 0) + break; + snd_iprintf(buffer, "tx %u:\n", stream); + snd_iprintf(buffer, " iso channel: %d\n", (int)buf.tx.iso); + snd_iprintf(buffer, " audio channels: %u\n", + buf.tx.number_audio); + snd_iprintf(buffer, " midi ports: %u\n", buf.tx.number_midi); + snd_iprintf(buffer, " speed: S%u\n", 100u << buf.tx.speed); + if (quadlets >= 68) { + dice_proc_fixup_string(buf.tx.names, TX_NAMES_SIZE); + snd_iprintf(buffer, " names: %s\n", buf.tx.names); + } + if (quadlets >= 70) { + snd_iprintf(buffer, " ac3 caps: %08x\n", + buf.tx.ac3_caps); + snd_iprintf(buffer, " ac3 enable: %08x\n", + buf.tx.ac3_enable); + } + } + + if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) + return; + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx)); + for (stream = 0; stream < tx_rx_header.number; ++stream) { + if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + + stream * tx_rx_header.size, + quadlets) < 0) + break; + snd_iprintf(buffer, "rx %u:\n", stream); + snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); + snd_iprintf(buffer, " sequence start: %u\n", + (int)buf.rx.seq_start); + snd_iprintf(buffer, " audio channels: %u\n", + buf.rx.number_audio); + snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); + if (quadlets >= 68) { + dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); + snd_iprintf(buffer, " names: %s\n", buf.rx.names); + } + if (quadlets >= 70) { + snd_iprintf(buffer, " ac3 caps: %08x\n", + buf.rx.ac3_caps); + snd_iprintf(buffer, " ac3 enable: %08x\n", + buf.rx.ac3_enable); + } + } + + quadlets = min_t(u32, sections[7], sizeof(buf.ext_sync) / 4); + if (quadlets >= 4) { + if (dice_proc_read_mem(dice, &buf.ext_sync, + sections[6], 4) < 0) + return; + snd_iprintf(buffer, "ext status:\n"); + snd_iprintf(buffer, " clock source: %s\n", + str_from_array(clock_sources, + ARRAY_SIZE(clock_sources), + buf.ext_sync.clock_source)); + snd_iprintf(buffer, " locked: %u\n", buf.ext_sync.locked); + snd_iprintf(buffer, " rate: %s\n", + str_from_array(rates, ARRAY_SIZE(rates), + buf.ext_sync.rate)); + snd_iprintf(buffer, " adat user data: "); + if (buf.ext_sync.adat_user_data & ADAT_USER_DATA_NO_DATA) + snd_iprintf(buffer, "-\n"); + else + snd_iprintf(buffer, "%x\n", + buf.ext_sync.adat_user_data); + } +} + +static void dice_create_proc(struct dice *dice) +{ + struct snd_info_entry *entry; + + if (!snd_card_proc_new(dice->card, "dice", &entry)) + snd_info_set_text_ops(entry, dice, dice_proc_read); +} + static void dice_card_free(struct snd_card *card) { struct dice *dice = card->private_data; @@ -1131,6 +1375,8 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) if (err < 0) goto error; + dice_create_proc(dice); + err = snd_card_register(card); if (err < 0) goto error; -- cgit From 61b8cf0222b256b4f793d99c8bdc9b216d067a76 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 8 Jan 2012 22:18:00 +0100 Subject: ALSA: dice: document quadlet alignment Doing accesses without quadlet alignment is a bad idea because the firmware's byte-swapping would garble the data; clarify this in the documentation. Signed-off-by: Clemens Ladisch --- sound/firewire/dice-interface.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/firewire/dice-interface.h b/sound/firewire/dice-interface.h index af916b9d07a4..27b044f84c81 100644 --- a/sound/firewire/dice-interface.h +++ b/sound/firewire/dice-interface.h @@ -7,9 +7,9 @@ /* * Generally, all registers can be read like memory, i.e., with quadlet read or - * block read transactions with any alignment or length. Writes are not - * allowed except where noted; quadlet-sized registers must be written with - * a quadlet write transaction. + * block read transactions with at least quadlet-aligned offset and length. + * Writes are not allowed except where noted; quadlet-sized registers must be + * written with a quadlet write transaction. * * All values are in big endian. The DICE firmware runs on a little-endian CPU * and just byte-swaps _all_ quadlets on the bus, so values without endianness -- cgit From ed7e48264cfd3b000ab8dd100e6aa4c1447dd93a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 22 Jan 2012 16:46:23 +0100 Subject: ALSA: dice: dice_proc_read: remove wrong typecast Remove a wrong typecast that resulted from a copy-and-paste error. Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 25a96362e1aa..5f0f102cf5f9 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -1053,8 +1053,7 @@ static void dice_proc_read(struct snd_info_entry *entry, break; snd_iprintf(buffer, "rx %u:\n", stream); snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); - snd_iprintf(buffer, " sequence start: %u\n", - (int)buf.rx.seq_start); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); snd_iprintf(buffer, " audio channels: %u\n", buf.rx.number_audio); snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); -- cgit From a471fcde8c2c4b65f110bb4210af3513ee4deeb8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 13 Feb 2012 21:55:13 +0100 Subject: ALSA: dice: fix detection of Weiss devices While most DICE devices keep TCAT's default category ID of 0x04, Weiss devices identify themselves with 0x00. Reported-by: Rolf Anderegg Signed-off-by: Clemens Ladisch --- sound/firewire/dice.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 5f0f102cf5f9..49d630ba2d9f 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -1109,7 +1109,10 @@ static void dice_card_free(struct snd_card *card) mutex_destroy(&dice->mutex); } -#define DICE_CATEGORY_ID 0x04 +#define OUI_WEISS 0x001c6a + +#define DICE_CATEGORY_ID 0x04 +#define WEISS_CATEGORY_ID 0x00 static int dice_interface_check(struct fw_unit *unit) { @@ -1123,15 +1126,15 @@ static int dice_interface_check(struct fw_unit *unit) struct fw_device *device = fw_parent_device(unit); struct fw_csr_iterator it; int key, value, vendor = -1, model = -1, err; - unsigned int i; + unsigned int category, i; __be32 pointers[ARRAY_SIZE(min_values)]; __be32 version; /* * Check that GUID and unit directory are constructed according to DICE * rules, i.e., that the specifier ID is the GUID's OUI, and that the - * GUID chip ID consists of the 8-bit DICE category ID, the 10-bit - * product ID, and a 22-bit serial number. + * GUID chip ID consists of the 8-bit category ID, the 10-bit product + * ID, and a 22-bit serial number. */ fw_csr_iterator_init(&it, unit->directory); while (fw_csr_iterator_next(&it, &key, &value)) { @@ -1144,7 +1147,11 @@ static int dice_interface_check(struct fw_unit *unit) break; } } - if (device->config_rom[3] != ((vendor << 8) | DICE_CATEGORY_ID) || + if (vendor == OUI_WEISS) + category = WEISS_CATEGORY_ID; + else + category = DICE_CATEGORY_ID; + if (device->config_rom[3] != ((vendor << 8) | category) || device->config_rom[4] >> 22 != model) return -ENODEV; -- cgit From b20be8de1b3972ccf9af72850b045214faa8d830 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 15 Oct 2013 20:26:05 +0200 Subject: ALSA: dice: restrict the driver to playback-only devices At the moment, this driver supports only playback, while FFADO supports (only) full-duplex devices. So, prevent conflicts by not claiming devices that would be better handled by FFADO. Signed-off-by: Clemens Ladisch --- sound/firewire/Kconfig | 10 ++++++---- sound/firewire/dice.c | 9 +++++++++ 2 files changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 915330989412..b3e274fe4a77 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -12,14 +12,16 @@ config SND_FIREWIRE_LIB depends on SND_PCM config SND_DICE - tristate "DICE devices (EXPERIMENTAL)" + tristate "DICE-based DACs (EXPERIMENTAL)" select SND_HWDEP select SND_PCM select SND_FIREWIRE_LIB help - Say Y here to include support for many FireWire audio interfaces - based on the DICE chip family (DICE-II/Jr/Mini) from TC Applied - Technologies. + Say Y here to include support for many DACs based on the DICE + chip family (DICE-II/Jr/Mini) from TC Applied Technologies. + + At the moment, this driver supports playback only. If you + want to use devices that support capturing, use FFADO instead. To compile this driver as a module, choose M here: the module will be called snd-dice. diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 49d630ba2d9f..6feee6614193 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -1128,6 +1128,7 @@ static int dice_interface_check(struct fw_unit *unit) int key, value, vendor = -1, model = -1, err; unsigned int category, i; __be32 pointers[ARRAY_SIZE(min_values)]; + __be32 tx_data[4]; __be32 version; /* @@ -1171,6 +1172,14 @@ static int dice_interface_check(struct fw_unit *unit) return -ENODEV; } + /* We support playback only. Let capture devices be handled by FFADO. */ + err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[2]) * 4, + tx_data, sizeof(tx_data), 0); + if (err < 0 || (tx_data[0] && tx_data[3])) + return -ENODEV; + /* * Check that the implemented DICE driver specification major version * number matches. -- cgit From cfcff69af8447df8dd3c5b14349c3b84b8b569a5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 20 Oct 2013 18:14:20 +0100 Subject: ASoC: si476x: Fix locking of core The conversion of the si476x to regmap removed locking of the core during register updates, allowing things like power state changes for the MFD to happen during a register update. Avoid this by taking the core lock in the DAI operations (which are the only things that do register updates) as we used to do in the open coded register I/O functions. Signed-off-by: Mark Brown Acked-by: Andrey Smirnov --- sound/soc/codecs/si476x.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 03645ce42063..52e7cb08434b 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -73,6 +73,7 @@ static const struct snd_soc_dapm_route si476x_dapm_routes[] = { static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { + struct si476x_core *core = i2c_mfd_cell_to_core(codec_dai->dev); int err; u16 format = 0; @@ -136,9 +137,14 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_update_bits(codec_dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK, format); + + si476x_core_unlock(core); + if (err < 0) { dev_err(codec_dai->codec->dev, "Failed to set output format\n"); return err; @@ -151,6 +157,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct si476x_core *core = i2c_mfd_cell_to_core(dai->dev); int rate, width, err; rate = params_rate(params); @@ -176,11 +183,13 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_write(dai->codec, SI476X_DIGITAL_IO_OUTPUT_SAMPLE_RATE, rate); if (err < 0) { dev_err(dai->codec->dev, "Failed to set sample rate\n"); - return err; + goto out; } err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, @@ -189,10 +198,13 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)); if (err < 0) { dev_err(dai->codec->dev, "Failed to set output width\n"); - return err; + goto out; } - return 0; +out: + si476x_core_unlock(core); + + return err; } static int si476x_codec_probe(struct snd_soc_codec *codec) -- cgit From cc1a95d9f6423ced191b6f264e9657d98844ea0d Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Sun, 20 Oct 2013 23:03:31 -0400 Subject: ALSA: hda - add codec ID for Valleyview2 display codec This patch adds codec ID (0x80862882) and module alias for Valleyview2 display codec. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 804adb872e63..fbd18599b0f9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2779,6 +2779,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -2833,6 +2834,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862880"); +MODULE_ALIAS("snd-hda-codec-id:80862882"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_LICENSE("GPL"); -- cgit From b55447a7301b12d509df4b2909ed38d125ad83d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Oct 2013 16:31:45 +0200 Subject: ALSA: hda - Fix typos in patch_hdmi.c ... which was introduced by the previous commit a4e9a38b, causing build errors without CONFIG_PROC_FS. Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index fbd18599b0f9..0f9b10322b2a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -539,12 +539,12 @@ static void eld_proc_free(struct hdmi_spec_per_pin *per_pin) } } #else -static inline int snd_hda_eld_proc_new(struct hdmi_spec_per_pin *per_pin, - int index) +static inline int eld_proc_new(struct hdmi_spec_per_pin *per_pin, + int index) { return 0; } -static inline void snd_hda_eld_proc_free(struct hdmi_spec_per_pin *per_pin) +static inline void eld_proc_free(struct hdmi_spec_per_pin *per_pin) { } #endif -- cgit From 30a765d6433413c0eba90c969eecf12dfa2d111a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 21 Oct 2013 19:07:34 +0530 Subject: ASoC: dont call dapm_sync while reporting jack always While reporting the jack status snd_soc_jack_report() invokes snd_soc_dapm_sync() always. This should be required when we have pins associated with jack and reporting enables or disables these. So add a check for this case Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 71358e3b54d9..23d43dac91da 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -65,6 +65,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; + unsigned int sync = 0; int enable; trace_snd_soc_jack_report(jack, mask, status); @@ -92,12 +93,16 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_enable_pin(dapm, pin->pin); else snd_soc_dapm_disable_pin(dapm, pin->pin); + + /* we need to sync for this case only */ + sync = 1; } /* Report before the DAPM sync to help users updating micbias status */ blocking_notifier_call_chain(&jack->notifier, jack->status, jack); - snd_soc_dapm_sync(dapm); + if (sync) + snd_soc_dapm_sync(dapm); snd_jack_report(jack->jack, jack->status); -- cgit From 75b9b65ee5a80e99efe2fd551d08bc86f115550f Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Mon, 21 Oct 2013 10:50:49 +0200 Subject: ASoC: kirkwood: add S/PDIF support This patch adds S/PDIF input/output for mvebu DT boards. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 99 +++++++++++++++++++++++++++++------- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- 3 files changed, 84 insertions(+), 19 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d0504a2d8c63..9ec38d15df9e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,9 +421,29 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; - -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -422,10 +459,34 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -443,6 +504,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +514,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -524,7 +586,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -535,6 +597,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e97164..65f2a5b9ec3b 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0c4856..d213832b0c72 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", -- cgit From 256ba181cb2ddeef8e0a9b0540b09e0f77bf5540 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 18 Oct 2013 18:37:42 +0300 Subject: ASoC: davinci-mcasp: Add location for data port registers to DT This patch adds a separate register location for data port registers to mcasp DT bindings. On am33xx SoCs the McASP registers are mapped trough L4 interconnect, but data port registers are also mapped trough L3 bus to a different memory location. Signed-off-by: Hebbar, Gururaja Signed-off-by: Darren Etheridge Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 8 ++- sound/soc/davinci/davinci-mcasp.c | 61 +++++++++++++++------- 2 files changed, 47 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 374e145c2ef1..c2ab8697e24a 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -6,7 +6,11 @@ Required properties: "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) -- reg : Should contain McASP registers offset and length +- reg : Should contain reg specifiers for the entries in the reg-names property. +- reg-names : Should contain: + * "mpu" for the main registers (required). For compatibility with + existing software, it is recommended this is the first entry. + * "dat" for separate data port register access (optional). - interrupts : Interrupt number for McASP - op-mode : I2S/DIT ops mode. - tdm-slots : Slots for TDM operation. @@ -15,7 +19,6 @@ Required properties: to "num-serializer" parameter. Each entry is a number indication serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) - Optional properties: - ti,hwmods : Must be "mcasp", n is controller instance starting 0 @@ -31,6 +34,7 @@ mcasp0: mcasp0@1d00000 { #address-cells = <1>; #size-cells = <0>; reg = <0x100000 0x3000>; + reg-names "mpu"; interrupts = <82 83>; op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <2>; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index cdfe959d6062..806bec34e4d9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1001,18 +1001,40 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { .name = "davinci-mcasp", }; +/* Some HW specific values and defaults. The rest is filled in from DT. */ +static struct snd_platform_data dm646x_mcasp_pdata = { + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_1, +}; + +static struct snd_platform_data da830_mcasp_pdata = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_2, +}; + +static struct snd_platform_data omap2_mcasp_pdata = { + .tx_dma_offset = 0, + .rx_dma_offset = 0, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_3, +}; + static const struct of_device_id mcasp_dt_ids[] = { { .compatible = "ti,dm646x-mcasp-audio", - .data = (void *)MCASP_VERSION_1, + .data = &dm646x_mcasp_pdata, }, { .compatible = "ti,da830-mcasp-audio", - .data = (void *)MCASP_VERSION_2, + .data = &da830_mcasp_pdata, }, { .compatible = "ti,omap2-mcasp-audio", - .data = (void *)MCASP_VERSION_3, + .data = &omap2_mcasp_pdata, }, { /* sentinel */ } }; @@ -1035,20 +1057,13 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata = pdev->dev.platform_data; return pdata; } else if (match) { - pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); - if (!pdata) { - ret = -ENOMEM; - goto nodata; - } + pdata = (struct snd_platform_data *) match->data; } else { /* control shouldn't reach here. something is wrong */ ret = -EINVAL; goto nodata; } - if (match->data) - pdata->version = (u8)((int)match->data); - ret = of_property_read_u32(np, "op-mode", &val); if (ret >= 0) pdata->op_mode = val; @@ -1124,7 +1139,7 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { struct davinci_pcm_dma_params *dma_data; - struct resource *mem, *ioarea, *res; + struct resource *mem, *ioarea, *res, *dat; struct snd_platform_data *pdata; struct davinci_audio_dev *dev; int ret; @@ -1145,10 +1160,15 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return -EINVAL; } - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!mem) { - dev_err(&pdev->dev, "no mem resource?\n"); - return -ENODEV; + dev_warn(dev->dev, + "\"mpu\" mem resource not found, using index 0\n"); + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } } ioarea = devm_request_mem_region(&pdev->dev, mem->start, @@ -1182,13 +1202,16 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->rxnumevt = pdata->rxnumevt; dev->dev = &pdev->dev; + dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); + if (!dat) + dat = mem; + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_playback; - dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + - mem->start); + dma_data->dma_addr = dat->start + pdata->tx_dma_offset; /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -1205,8 +1228,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_capture; - dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + - mem->start); + dma_data->dma_addr = dat->start + pdata->rx_dma_offset; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -1305,4 +1327,3 @@ module_platform_driver(davinci_mcasp_driver); MODULE_AUTHOR("Steve Chen"); MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface"); MODULE_LICENSE("GPL"); - -- cgit From 4023fe6ff2192d6050647571ea54f5497b2ec8f6 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 18 Oct 2013 18:37:43 +0300 Subject: ASoC: davinci-mcasp: Extract DMA channels directly from DT Extract DMA channels directly from DT as they can not be found from platform resources anymore. This is a work-around until davinci audio driver is updated to use dmaengine. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 5 +++ include/linux/platform_data/davinci_asp.h | 2 + sound/soc/davinci/davinci-mcasp.c | 45 +++++++++++++++------- 3 files changed, 38 insertions(+), 14 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index c2ab8697e24a..c3ccde71f97a 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -18,6 +18,11 @@ Required properties: - serial-dir : A list of serializer pin mode. The list number should be equal to "num-serializer" parameter. Each entry is a number indication serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) +- dmas: two element list of DMA controller phandles and DMA request line + ordered pairs. +- dma-names: identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. The dma + identifiers must be "rx" and "tx". Optional properties: diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 8db5ae03b6e3..689a856b86f9 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -84,6 +84,8 @@ struct snd_platform_data { u8 version; u8 txnumevt; u8 rxnumevt; + int tx_dma_channel; + int rx_dma_channel; }; enum { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 806bec34e4d9..4c207508348f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1047,6 +1047,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct snd_platform_data *pdata = NULL; const struct of_device_id *match = of_match_device(mcasp_dt_ids, &pdev->dev); + struct of_phandle_args dma_spec; const u32 *of_serial_dir32; u8 *of_serial_dir; @@ -1109,6 +1110,28 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->serial_dir = of_serial_dir; } + ret = of_property_match_string(np, "dma-names", "tx"); + if (ret < 0) + goto nodata; + + ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, + &dma_spec); + if (ret < 0) + goto nodata; + + pdata->tx_dma_channel = dma_spec.args[0]; + + ret = of_property_match_string(np, "dma-names", "rx"); + if (ret < 0) + goto nodata; + + ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, + &dma_spec); + if (ret < 0) + goto nodata; + + pdata->rx_dma_channel = dma_spec.args[0]; + ret = of_property_read_u32(np, "tx-num-evt", &val); if (ret >= 0) pdata->txnumevt = val; @@ -1213,15 +1236,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->sram_size = pdata->sram_size_playback; dma_data->dma_addr = dat->start + pdata->tx_dma_offset; - /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENODEV; - goto err_release_clk; - } - - dma_data->channel = res->start; + if (res) + dma_data->channel = res->start; + else + dma_data->channel = pdata->tx_dma_channel; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; @@ -1231,13 +1250,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->dma_addr = dat->start + pdata->rx_dma_offset; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENODEV; - goto err_release_clk; - } + if (res) + dma_data->channel = res->start; + else + dma_data->channel = pdata->rx_dma_channel; - dma_data->channel = res->start; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); -- cgit From 3af9e0315699b60762157662f721f50fd1fe529b Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 18 Oct 2013 18:37:44 +0300 Subject: ASoC: davinci-mcasp: Change compatible property model to more accurate Change the model omap2-mcasp-audio in compatible property to am33xx-mcasp-audio as omap2 does not have mcasp. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index c3ccde71f97a..1945aecf0a3a 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,7 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) + "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, TI81xx) - reg : Should contain reg specifiers for the entries in the reg-names property. - reg-names : Should contain: diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4c207508348f..bbc9a0793eb9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1033,7 +1033,7 @@ static const struct of_device_id mcasp_dt_ids[] = { .data = &da830_mcasp_pdata, }, { - .compatible = "ti,omap2-mcasp-audio", + .compatible = "ti,am33xx-mcasp-audio", .data = &omap2_mcasp_pdata, }, { /* sentinel */ } -- cgit From 62561b39ea346ec2e48e01de9fd6f38383b67bd3 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 18 Oct 2013 18:37:45 +0300 Subject: ASoC: davinci-mcasp: Improve DT bindings document Makes interrupts property optional as the interrupts are not currently used by the driver and adds interrupt-names property to name listed interrupts. Currently know interrupt names are "tx" and "rx". - Improve tdm-slots propery description - Improve op-mode property description - Add pinctrl-names and pinctrl-0 properties - Remove #address-cells and #size-cells as they are not needed. - Bracket named interrupts property tuples for uniformity. - Add missing "for" to serial-dir prop in DT bindings doc. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 24 ++++++++++++++-------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 1945aecf0a3a..b925bf955731 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -11,13 +11,14 @@ Required properties: * "mpu" for the main registers (required). For compatibility with existing software, it is recommended this is the first entry. * "dat" for separate data port register access (optional). -- interrupts : Interrupt number for McASP -- op-mode : I2S/DIT ops mode. -- tdm-slots : Slots for TDM operation. +- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, + IEC60958-1, and AES-3 formats. +- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted + or received over one serializer. - num-serializer : Serializers used by McASP. -- serial-dir : A list of serializer pin mode. The list number should be equal - to "num-serializer" parameter. Each entry is a number indication - serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) +- serial-dir : A list of serializer configuration. Each entry is a number + indication for serializer pin direction. + (0 - INACTIVE, 1 - TX, 2 - RX) - dmas: two element list of DMA controller phandles and DMA request line ordered pairs. - dma-names: identifier string for each DMA request line in the dmas property. @@ -31,16 +32,21 @@ Optional properties: - rx-num-evt : FIFO levels. - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture +- interrupts : Interrupt numbers for McASP, currently not used by the driver +- interrupt-names : Known interrupt names are "tx" and "rx" +- pinctrl-0: Should specify pin control group used for this controller. +- pinctrl-names: Should contain only one value - "default", for more details + please refer to pinctrl-bindings.txt + Example: mcasp0: mcasp0@1d00000 { compatible = "ti,da830-mcasp-audio"; - #address-cells = <1>; - #size-cells = <0>; reg = <0x100000 0x3000>; reg-names "mpu"; - interrupts = <82 83>; + interrupts = <82>, <83>; + interrupts-names = "tx", "rx"; op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <2>; num-serializer = <16>; -- cgit From 1427e660b49e87cd842dba94158b0fc73030c17e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 18 Oct 2013 18:37:46 +0300 Subject: ASoC: davinci-mcasp: Remove redundant num-serializer DT parameter The serial-dir array gives this information so there is no need to have the num-serializer property in DT description. Just ignore the property in the driver the DTS files can be updated separately without regression. Update the documentation at the same time for davinci-mcasp Signed-off-by: Peter Ujfalusi Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 1 - sound/soc/davinci/davinci-mcasp.c | 22 +++++----------------- 2 files changed, 5 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index b925bf955731..0aa416b68e3d 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -49,7 +49,6 @@ mcasp0: mcasp0@1d00000 { interrupts-names = "tx", "rx"; op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <2>; - num-serializer = <16>; serial-dir = < 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 0 0 diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index bbc9a0793eb9..71e14bb3a8cd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1050,7 +1050,6 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct of_phandle_args dma_spec; const u32 *of_serial_dir32; - u8 *of_serial_dir; u32 val; int i, ret = 0; @@ -1081,32 +1080,21 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->tdm_slots = val; } - ret = of_property_read_u32(np, "num-serializer", &val); - if (ret >= 0) - pdata->num_serializer = val; - of_serial_dir32 = of_get_property(np, "serial-dir", &val); val /= sizeof(u32); - if (val != pdata->num_serializer) { - dev_err(&pdev->dev, - "num-serializer(%d) != serial-dir size(%d)\n", - pdata->num_serializer, val); - ret = -EINVAL; - goto nodata; - } - if (of_serial_dir32) { - of_serial_dir = devm_kzalloc(&pdev->dev, - (sizeof(*of_serial_dir) * val), - GFP_KERNEL); + u8 *of_serial_dir = devm_kzalloc(&pdev->dev, + (sizeof(*of_serial_dir) * val), + GFP_KERNEL); if (!of_serial_dir) { ret = -ENOMEM; goto nodata; } - for (i = 0; i < pdata->num_serializer; i++) + for (i = 0; i < val; i++) of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]); + pdata->num_serializer = val; pdata->serial_dir = of_serial_dir; } -- cgit From d7711dc5930ced241c4f6e9b14df2a92814f9f12 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 14:13:04 +0100 Subject: ASoC: ep93xx: Open code dma channel request Currently the ep93xx DMA code is one of the few users relying on the fact that the compat code uses the dma_data as the filter data for non-DT channel requests. Since the rest of the core expects this to be a struct snd_dmaengine_dai_data this isn't terribly helpful this will be changed to use the already existing filter data so avoid breaking ep93xx by open coding the current behaviour. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/cirrus/ep93xx-pcm.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 0e9f56e0d4b2..cfe517e68009 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -57,9 +57,22 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) return false; } +static struct dma_chan *ep93xx_compat_request_channel( + struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + return snd_dmaengine_pcm_request_channel(ep93xx_pcm_dma_filter, + dma_data); +} + static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .pcm_hardware = &ep93xx_pcm_hardware, .compat_filter_fn = ep93xx_pcm_dma_filter, + .compat_request_channel = ep93xx_compat_request_channel, .prealloc_buffer_size = 131072, }; -- cgit From 0eef5381b7271702c7e65c637cb46804c482a90a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 14:17:03 +0100 Subject: ASoC: tegra: Remove redundant initialisation of compat_filter_fn Setting a field in a static struct to NULL has no effect so don't bother (and don't generate false positives for grep). Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen Acked-by: Stephen Warren --- sound/soc/tegra/tegra_pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index f056f632557c..7b2d23ba69b3 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -56,7 +56,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { static const struct snd_dmaengine_pcm_config tegra_dmaengine_pcm_config = { .pcm_hardware = &tegra_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = NULL, .prealloc_buffer_size = PAGE_SIZE * 8, }; -- cgit From d79e07c95d1328773509c69131e5f25cac5dbf50 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Wed, 23 Oct 2013 15:30:13 +0300 Subject: ASoC: davinci: Add support for AM33xx SoC Audio AM33xx uses same McASP IP as the Davinci Platform. This patch updates Kconfig and makefile to enable build for McASP, PCM & Codec drivers. Signed-off-by: Hebbar, Gururaja Signed-off-by: Darren Etheridge Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 18 +++++++++++++++--- sound/soc/davinci/Makefile | 1 + 2 files changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index c82f89c9475b..95970f5db3ec 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,9 +1,10 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for the TI DAVINCI chip" - depends on ARCH_DAVINCI + tristate "SoC Audio for the TI DAVINCI or AM33XX chip" + depends on ARCH_DAVINCI || SOC_AM33XX help + Platform driver for daVinci or AM33xx Say Y or M if you want to add support for codecs attached to - the DAVINCI AC97 or I2S interface. You will also need + the DAVINCI AC97, I2S, or McASP interface. You will also need to select the audio interfaces to support below. config SND_DAVINCI_SOC_I2S @@ -15,6 +16,17 @@ config SND_DAVINCI_SOC_MCASP config SND_DAVINCI_SOC_VCIF tristate +config SND_AM33XX_SOC_EVM + tristate "SoC Audio for the AM33XX chip based boards" + depends on SND_DAVINCI_SOC && SOC_AM33XX + select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_MCASP + help + Say Y or M if you want to add support for SoC audio on AM33XX + boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, + AM335X-EVMSK, and BeagelBone with AudioCape boards have this + setup. + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a396ab6d6d5e..bc81e79fc301 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o snd-soc-evm-objs := davinci-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_AM33XX_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o -- cgit From d5faaa34262d432130f07fed958c8161ef2715ab Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 23 Oct 2013 15:30:15 +0300 Subject: ASoC: davinci-mcasp: Remove last reference to num-serializer in DT doc Remove last reference to num-serializer in davinci-mcasp devicetree binding document. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 0aa416b68e3d..ed785b3f67be 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -15,7 +15,6 @@ Required properties: IEC60958-1, and AES-3 formats. - tdm-slots : Slots for TDM operation. Indicates number of channels transmitted or received over one serializer. -- num-serializer : Serializers used by McASP. - serial-dir : A list of serializer configuration. Each entry is a number indication for serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) -- cgit From ee2f615d6e59cea2b9a415661a7f27caffcb3528 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Wed, 23 Oct 2013 15:30:14 +0300 Subject: ASoC: davinci-evm: Add device tree binding Device tree support for Davinci Machine driver When the board boots with device tree, the driver will receive card, codec, dai interface details (like the card name, DAPM routing map, phandle for the audio components described in the dts file, codec mclk speed). The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. Signed-off-by: Hebbar, Gururaja Signed-off-by: Darren Etheridge Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../bindings/sound/davinci-evm-audio.txt | 42 +++++++ sound/soc/davinci/davinci-evm.c | 124 ++++++++++++++++++++- 2 files changed, 164 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 000000000000..865178d5cdf3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -0,0 +1,42 @@ +* Texas Instruments SoC audio setups with TLV320AIC3X Codec + +Required properties: +- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx +- ti,model : The user-visible name of this sound complex. +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,mcasp-controller : The phandle of the McASP controller +- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec +- ti,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the codec's pins, and the jacks on the board: + + Board connectors: + + * Headphone Jack + * Line Out + * Mic Jack + * Line In + + +Example: + +sound { + compatible = "ti,da830-evm-audio"; + ti,model = "DA830 EVM"; + ti,audio-codec = <&tlv320aic3x>; + ti,mcasp-controller = <&mcasp1>; + ti,codec-clock-rate = <12000000>; + ti,audio-routing = + "Headphone Jack", "HPLOUT", + "Headphone Jack", "HPROUT", + "Line Out", "LLOUT", + "Line Out", "RLOUT", + "MIC3L", "Mic Bias 2V", + "MIC3R", "Mic Bias 2V", + "Mic Bias 2V", "Mic Jack", + "LINE1L", "Line In", + "LINE2L", "Line In", + "LINE1R", "Line In", + "LINE2R", "Line In"; +}; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 2f8161c1d5f0..623eb5e7c089 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -23,6 +24,8 @@ #include #include +#include + #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" @@ -121,13 +124,22 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + int ret; /* Add davinci-evm specific widgets */ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + if (np) { + ret = snd_soc_of_parse_audio_routing(codec->card, + "ti,audio-routing"); + if (ret) + return ret; + } else { + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + } /* not connected */ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); @@ -312,6 +324,98 @@ static struct snd_soc_card da850_snd_soc_card = { .drvdata = &da850_snd_soc_card_drvdata, }; +#if defined(CONFIG_OF) + +/* + * The struct is used as place holder. It will be completely + * filled with data from dt node. + */ +static struct snd_soc_dai_link evm_dai_tlv320aic3x = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .codec_dai_name = "tlv320aic3x-hifi", + .ops = &evm_ops, + .init = evm_aic3x_init, +}; + +static const struct of_device_id davinci_evm_dt_ids[] = { + { + .compatible = "ti,da830-evm-audio", + .data = (void *) &evm_dai_tlv320aic3x, + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, davinci_evm_dt_ids); + +/* davinci evm audio machine driver */ +static struct snd_soc_card evm_soc_card = { + .owner = THIS_MODULE, + .num_links = 1, +}; + +static int davinci_evm_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + const struct of_device_id *match = + of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev); + struct snd_soc_dai_link *dai = (struct snd_soc_dai_link *) match->data; + struct snd_soc_card_drvdata_davinci *drvdata = NULL; + int ret = 0; + + evm_soc_card.dai_link = dai; + + dai->codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0); + if (!dai->codec_of_node) + return -EINVAL; + + dai->cpu_of_node = of_parse_phandle(np, "ti,mcasp-controller", 0); + if (!dai->cpu_of_node) + return -EINVAL; + + dai->platform_of_node = dai->cpu_of_node; + + evm_soc_card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); + if (ret) + return ret; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + + ret = of_property_read_u32(np, "ti,codec-clock-rate", &drvdata->sysclk); + if (ret < 0) + return -EINVAL; + + snd_soc_card_set_drvdata(&evm_soc_card, drvdata); + ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card); + + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + + return ret; +} + +static int davinci_evm_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver davinci_evm_driver = { + .probe = davinci_evm_probe, + .remove = davinci_evm_remove, + .driver = { + .name = "davinci_evm", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(davinci_evm_dt_ids), + }, +}; +#endif + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -320,6 +424,15 @@ static int __init evm_init(void) int index; int ret; + /* + * If dtb is there, the devices will be created dynamically. + * Only register platfrom driver structure. + */ +#if defined(CONFIG_OF) + if (of_have_populated_dt()) + return platform_driver_register(&davinci_evm_driver); +#endif + if (machine_is_davinci_evm()) { evm_snd_dev_data = &dm6446_snd_soc_card_evm; index = 0; @@ -355,6 +468,13 @@ static int __init evm_init(void) static void __exit evm_exit(void) { +#if defined(CONFIG_OF) + if (of_have_populated_dt()) { + platform_driver_unregister(&davinci_evm_driver); + return; + } +#endif + platform_device_unregister(evm_snd_device); } -- cgit From f95a48834cb9c581eec952215666a323136f339f Mon Sep 17 00:00:00 2001 From: Sebastian Reichel Date: Wed, 23 Oct 2013 14:03:28 +0200 Subject: ASoC: tpa6130a2: Add device tree support Add device tree support to tpa6130a2 driver and document the bindings. Signed-off-by: Sebastian Reichel Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tpa6130a2.txt | 27 ++++++++++++++++++ sound/soc/codecs/tpa6130a2.c | 32 ++++++++++++++++------ 2 files changed, 50 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt new file mode 100644 index 000000000000..6dfa740e4b2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tpa6130a2.txt @@ -0,0 +1,27 @@ +Texas Instruments - tpa6130a2 Codec module + +The tpa6130a2 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tpa6130a2" - TPA6130A2 + "ti,tpa6140a2" - TPA6140A2 + + +- reg - - I2C slave address + +- Vdd-supply - - power supply regulator + +Optional properties: + +- power-gpio - gpio pin to power the device + +Example: + +tpa6130a2: tpa6130a2@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index c58bee8346ce..998555f2a8aa 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -30,6 +30,7 @@ #include #include #include +#include #include "tpa6130a2.h" @@ -364,30 +365,33 @@ static int tpa6130a2_probe(struct i2c_client *client, { struct device *dev; struct tpa6130a2_data *data; - struct tpa6130a2_platform_data *pdata; + struct tpa6130a2_platform_data *pdata = client->dev.platform_data; + struct device_node *np = client->dev.of_node; const char *regulator; int ret; dev = &client->dev; - if (client->dev.platform_data == NULL) { - dev_err(dev, "Platform data not set\n"); - dump_stack(); - return -ENODEV; - } - data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); if (data == NULL) { dev_err(dev, "Can not allocate memory\n"); return -ENOMEM; } + if (pdata) { + data->power_gpio = pdata->power_gpio; + } else if (np) { + data->power_gpio = of_get_named_gpio(np, "power-gpio", 0); + } else { + dev_err(dev, "Platform data not set\n"); + dump_stack(); + return -ENODEV; + } + tpa6130a2_client = client; i2c_set_clientdata(tpa6130a2_client, data); - pdata = client->dev.platform_data; - data->power_gpio = pdata->power_gpio; data->id = id->driver_data; mutex_init(&data->mutex); @@ -466,10 +470,20 @@ static const struct i2c_device_id tpa6130a2_id[] = { }; MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tpa6130a2_of_match[] = { + { .compatible = "ti,tpa6130a2", }, + { .compatible = "ti,tpa6140a2" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tpa6130a2_of_match); +#endif + static struct i2c_driver tpa6130a2_i2c_driver = { .driver = { .name = "tpa6130a2", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tpa6130a2_of_match), }, .probe = tpa6130a2_probe, .remove = tpa6130a2_remove, -- cgit From 055032142c42d2821c4aa617915292d6a08d56fc Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 23 Oct 2013 11:47:43 +0800 Subject: ALSA: Add SoC on-chip internal ram support for DMA buffer allocation Now it's quite common that an SoC contains its on-chip internal RAM. By using this RAM space for DMA buffer during audio playback/record, we can shutdown the voltage for external RAM to save power. So add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen Reviewed-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 1 + sound/core/memalloc.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 6 ++++++ 3 files changed, 59 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b8213df7..510aec437f72 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f4fe0c..18c1d4759ccb 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include @@ -157,6 +158,46 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + * @size: number of bytes to allocate from the iram + * + * This function requires iram phandle provided via of_node + */ +void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) +{ + struct device *dev = dmab->dev.dev; + struct gen_pool *pool = NULL; + + if (dev->of_node) + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + + if (!pool) + return; + + /* Assign the pool into private_data field */ + dmab->private_data = pool; + + dmab->area = (void *)gen_pool_alloc(pool, size); + if (!dmab->area) + return; + + dmab->addr = gen_pool_virt_to_phys(pool, (unsigned long)dmab->area); +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + */ +void snd_free_dev_iram(struct snd_dma_buffer *dmab) +{ + struct gen_pool *pool = dmab->private_data; + + if (pool && dmab->area) + gen_pool_free(pool, (unsigned long)dmab->area, dmab->bytes); +} #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +238,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_malloc_dev_iram(dmab, size); + if (dmab->area) + break; + /* Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +318,9 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_free_dev_iram(dmab); + break; case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6d702c..513f0954d92e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,12 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- cgit From ea73b7ddf13548afd666373dc5e26ee7c812a3fe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 17:43:51 +0100 Subject: ASoC: dmaengine: Support custom channel names Some devices have more than just simple TX and RX DMA channels, for example modern Samsung I2S IPs support a secondary transmit DMA stream which is mixed into the primary stream during playback. Allow such devices to specify the names of the channels to be requested in their dma_data. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- include/sound/dmaengine_pcm.h | 6 ++++++ sound/soc/soc-generic-dmaengine-pcm.c | 13 ++++++++++++- 2 files changed, 18 insertions(+), 1 deletion(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 83b2c3e95a1a..15017311f2e9 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. + * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { @@ -69,6 +70,7 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + const char *chan_name; unsigned int fifo_size; }; @@ -98,6 +100,10 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) +/* + * The PCM streams have custom channel names specified. + */ +#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 99f9495c1c40..793cd6c246f6 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -183,6 +183,8 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; + struct device *dev = rtd->platform->dev; + struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -203,6 +205,13 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (!pcm->chan[i] && + (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) + pcm->chan[i] = dma_request_slave_channel(dev, + dma_data->chan_name); + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); @@ -275,7 +284,9 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, { unsigned int i; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) + if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || + !dev->of_node) return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { -- cgit From b7baa9a0e324e32433d894374c3bddea29961aa0 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Wed, 23 Oct 2013 12:14:54 -0700 Subject: ALSA: hda - Remove OOM message after input_allocate_device Emitting an OOM message isn't necessary after input_allocate_device as there's a generic OOM and a dump_stack already done. Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 63c99090a4ec..20d065a17226 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -151,10 +151,8 @@ static int snd_hda_do_attach(struct hda_beep *beep) int err; input_dev = input_allocate_device(); - if (!input_dev) { - printk(KERN_INFO "hda_beep: unable to allocate input device\n"); + if (!input_dev) return -ENOMEM; - } /* setup digital beep device */ input_dev->name = "HDA Digital PCBeep"; -- cgit From 1d04c9de5c76df113e4af7120feb53c628b5efcc Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 24 Oct 2013 11:35:18 +0200 Subject: ALSA: hda - Add support of ALC255 codecs It's just another variant of ALC269 & co. Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52c26d3a61d4..1c4f8c545fe7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2540,6 +2540,7 @@ enum { ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, + ALC269_TYPE_ALC255, }; /* @@ -2565,6 +2566,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC282: case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: + case ALC269_TYPE_ALC255: ssids = alc269_ssids; break; default: @@ -4102,6 +4104,9 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0286: spec->codec_variant = ALC269_TYPE_ALC286; break; + case 0x10ec0255: + spec->codec_variant = ALC269_TYPE_ALC255; + break; } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -4814,6 +4819,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, + { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, -- cgit From 161ebf295e2ad7876195143342362d2cf6ffaca2 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 24 Oct 2013 11:36:33 +0200 Subject: ALSA: hda - Add support of ALC285 / ALC293 codecs Yet another variants of ALC269. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c4f8c545fe7..3016467c04cc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2539,6 +2539,7 @@ enum { ALC269_TYPE_ALC282, ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, + ALC269_TYPE_ALC285, ALC269_TYPE_ALC286, ALC269_TYPE_ALC255, }; @@ -2559,6 +2560,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC269VC: case ALC269_TYPE_ALC280: case ALC269_TYPE_ALC284: + case ALC269_TYPE_ALC285: ssids = alc269va_ssids; break; case ALC269_TYPE_ALC269VB: @@ -4101,6 +4103,10 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0292: spec->codec_variant = ALC269_TYPE_ALC284; break; + case 0x10ec0285: + case 0x10ec0293: + spec->codec_variant = ALC269_TYPE_ALC285; + break; case 0x10ec0286: spec->codec_variant = ALC269_TYPE_ALC286; break; @@ -4833,9 +4839,11 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 }, { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 }, + { .id = 0x10ec0285, .name = "ALC285", .patch = patch_alc269 }, { .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 }, { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 }, { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 }, + { .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- cgit From 95f74c41b2e53f541f2f66c8ba3dac1601ebd409 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Oct 2013 01:24:15 +0200 Subject: ALSA: hda - Fix mute LED on HP laptops in runtime suspend When HP laptops with mute and mic-record LEDs go to runtime suspend, these LEDs are turned on forcibly no matter whether GPIO pis are on or off. This strange behavior seems triggered by resetting the HD-audio bus link at azx_rutime_suspend(). So, just add a new hda_bus flag to avoid the link reset at runtime suspend and set it for these HP machines. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 3 ++- sound/pci/hda/patch_sigmatel.c | 4 +++- 3 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7aa9870040c1..77db69480c19 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -698,6 +698,7 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ + unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */ int primary_dig_out_type; /* primary digital out PCM type */ }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e61a019aa5e..a0a06f79daa3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2986,7 +2986,8 @@ static int azx_runtime_suspend(struct device *dev) STATESTS_INT_MASK); azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!chip->bus->avoid_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fba0cef1c47f..69a549a82345 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2091,8 +2091,10 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ + codec->bus->avoid_link_reset = 1; + } } static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, -- cgit From 90130d2e8f75c7181cef514e8a1491925f386a16 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 21:38:26 +0100 Subject: ASoC: dmaengine: Use filter_data rather than dma_data for compat requests When using the legacy filter function channel requests we currently pass the audio specific struct snd_dmaengine_dai_dma_data which isn't likely to be helpful for actual filtering. Since there's already a field in the structure called filter_data clearly intended for use here convert the driver to use that. All existing users of plain filter functions have been converted to use an explicit compat function to override this behaviour except i.MX which is working around this issue in its filter function and is updated to just use filter_data directly here. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/fsl/imx-pcm-dma.c | 4 +--- sound/soc/soc-generic-dmaengine-pcm.c | 5 ++++- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 4dc1296688e9..aee23077080a 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -25,12 +25,10 @@ static bool filter(struct dma_chan *chan, void *param) { - struct snd_dmaengine_dai_dma_data *dma_data = param; - if (!imx_dma_is_general_purpose(chan)) return false; - chan->private = dma_data->filter_data; + chan->private = param; return true; } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 793cd6c246f6..0c469cbbe881 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -168,6 +168,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_pcm_substream *substream) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; @@ -176,7 +179,7 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return pcm->config->compat_request_channel(rtd, substream); return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); + dma_data->filter_data); } static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) -- cgit From 1abe729f783fece81d93e9a0253fd8079f19d7f6 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 24 Oct 2013 18:15:29 +0800 Subject: ASoC: fsl: Add missing pm to current machine drivers Add missing pm to current machine drivers so that all of them would correctly do suspend/resume. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-sgtl5000.c | 1 + sound/soc/fsl/imx-wm8962.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index a2fd7321b5a9..79cee782dbbf 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,6 +160,7 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index ca1be1d9dcf0..73da709069df 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -202,6 +202,7 @@ static struct platform_driver imx_sgtl5000_driver = { .driver = { .name = "imx-sgtl5000", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 722afe69169e..8e5b2c6a16d9 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -311,6 +311,7 @@ static struct platform_driver imx_wm8962_driver = { .driver = { .name = "imx-wm8962", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, -- cgit From 154bab12e1db824ad12ffd6df2d213dfb3d09878 Mon Sep 17 00:00:00 2001 From: Denis Carikli Date: Thu, 24 Oct 2013 14:13:48 +0200 Subject: ASoC: eukrea-tlv320: Use dev_err instead of pr_err. It also contains a minor style cleanup. Signed-off-by: Denis Carikli Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca2c1de..5983740be123 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -42,7 +42,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the cpu dai format.\n"); return ret; } @@ -50,14 +51,16 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec format.\n"); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec sysclk.\n"); return ret; } snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); @@ -65,7 +68,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + dev_err(cpu_dai->dev, + "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; } @@ -155,7 +159,8 @@ static struct platform_driver eukrea_tlv320_driver = { .owner = THIS_MODULE, }, .probe = eukrea_tlv320_probe, - .remove = eukrea_tlv320_remove,}; + .remove = eukrea_tlv320_remove, +}; module_platform_driver(eukrea_tlv320_driver); -- cgit From a5606f85611267047206d8ba055bc0e4ba166ad3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Oct 2013 14:25:32 +0200 Subject: ALSA: Add ifdef CONFIG_GENERIC_ALLOCATOR for SNDRV_DMA_TYPE_IRAM code It turned out that we can't use gen_pool_*() functions on archs without CONFIG_GENERIC_ALLOCATOR (resulting in missing symbols), since linux/genalloc.h doesn't provide dummy functions for all. We'd be able to fix linux/genalloc.h size, but I take an easier path for now... Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 4 ++++ sound/core/memalloc.c | 4 ++++ sound/core/pcm_native.c | 2 ++ 3 files changed, 10 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 510aec437f72..af9983970417 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,7 +52,11 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV +#endif /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 18c1d4759ccb..51a79218815b 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -238,6 +238,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA +#ifdef CONFIG_GENERIC_ALLOCATOR case SNDRV_DMA_TYPE_DEV_IRAM: snd_malloc_dev_iram(dmab, size); if (dmab->area) @@ -246,6 +247,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, * so if we fail to malloc, try to fetch memory traditionally. */ dmab->dev.type = SNDRV_DMA_TYPE_DEV; +#endif /* CONFIG_GENERIC_ALLOCATOR */ case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -318,9 +320,11 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA +#ifdef CONFIG_GENERIC_ALLOCATOR case SNDRV_DMA_TYPE_DEV_IRAM: snd_free_dev_iram(dmab); break; +#endif /* CONFIG_GENERIC_ALLOCATOR */ case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 513f0954d92e..b71be579c6ec 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,12 +3199,14 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; +#ifdef CONFIG_GENERIC_ALLOCATOR if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); return remap_pfn_range(area, area->vm_start, substream->dma_buffer.addr >> PAGE_SHIFT, area->vm_end - area->vm_start, area->vm_page_prot); } +#endif /* CONFIG_GENERIC_ALLOCATOR */ #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- cgit From 917f4b5cba78980a527098a910d94139d3e82c8d Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 24 Oct 2013 16:37:31 +0530 Subject: ALSA: compress: fix drain calls blocking other compress functions The drain and drain_notify callback were blocked by low level driver untill the draining was complete. Due to this being invoked with big fat mutex held, others ops like reading timestamp, calling pause, drop were blocked. So to fix this we add a new snd_compr_drain_notify() API. This would be required to be invoked by low level driver when drain or partial drain has been completed by the DSP. Thus we make the drain and partial_drain callback as non blocking and driver returns immediately after notifying DSP. The waiting is done while relasing the lock so that other ops can go ahead. Signed-off-by: Vinod Koul CC: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 12 ++++++++++++ sound/core/compress_offload.c | 41 ++++++++++++++++++++++++++++++++++++++--- 2 files changed, 50 insertions(+), 3 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 9031a26249b5..175ab3237b58 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -48,6 +48,8 @@ struct snd_compr_ops; * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep + * @wait: drain wait queue + * @drain_wake: condition for drain wake */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -59,6 +61,8 @@ struct snd_compr_runtime { u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; + wait_queue_head_t wait; + unsigned int drain_wake; void *private_data; }; @@ -171,4 +175,12 @@ static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) wake_up(&stream->runtime->sleep); } +static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) +{ + snd_BUG_ON(!stream); + + stream->runtime->drain_wake = 1; + wake_up(&stream->runtime->wait); +} + #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index bea523a5d852..3eb47d0006a7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -123,6 +123,7 @@ static int snd_compr_open(struct inode *inode, struct file *f) } runtime->state = SNDRV_PCM_STATE_OPEN; init_waitqueue_head(&runtime->sleep); + init_waitqueue_head(&runtime->wait); data->stream.runtime = runtime; f->private_data = (void *)data; mutex_lock(&compr->lock); @@ -682,12 +683,34 @@ static int snd_compr_stop(struct snd_compr_stream *stream) if (!retval) { stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); + snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; } return retval; } +static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) +{ + /* + * We are called with lock held. So drop the lock while we wait for + * drain complete notfication from the driver + * + * It is expected that driver will notify the drain completion and then + * stream will be moved to SETUP state, even if draining resulted in an + * error. We can trigger next track after this. + */ + stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + mutex_unlock(&stream->device->lock); + + wait_event(stream->runtime->wait, stream->runtime->drain_wake); + + wake_up(&stream->runtime->sleep); + mutex_lock(&stream->device->lock); + + return 0; +} + static int snd_compr_drain(struct snd_compr_stream *stream) { int retval; @@ -695,11 +718,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream) if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || stream->runtime->state == SNDRV_PCM_STATE_SETUP) return -EPERM; + + stream->runtime->drain_wake = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); - if (!retval) { - stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + if (retval) { + pr_err("SND_COMPR_TRIGGER_DRAIN failed %d\n", retval); wake_up(&stream->runtime->sleep); + return retval; } + + retval = snd_compress_wait_for_drain(stream); + stream->runtime->state = SNDRV_PCM_STATE_SETUP; return retval; } @@ -735,10 +764,16 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) if (stream->next_track == false) return -EPERM; + stream->runtime->drain_wake = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) { + pr_err("Partial drain returned failure\n"); + wake_up(&stream->runtime->sleep); + return retval; + } stream->next_track = false; - return retval; + return snd_compress_wait_for_drain(stream); } static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) -- cgit From 88011c0911ee7a1b001393662b9884fb7f32eec6 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 24 Oct 2013 15:45:24 +0200 Subject: ALSA: hda/realtek - Raise the delay for alc283_shutup Some machine with 85ms delay might be happen pop noise when codec enter to D3. Raise up to 100ms delay will be match for more machine. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3016467c04cc..dde858be3be5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2656,7 +2656,7 @@ static void alc283_shutup(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); if (hp_pin_sense) - msleep(85); + msleep(100); snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); @@ -2665,7 +2665,7 @@ static void alc283_shutup(struct hda_codec *codec) alc_write_coef_idx(codec, 0x46, val | (3 << 12)); if (hp_pin_sense) - msleep(85); + msleep(100); snd_hda_shutup_pins(codec); alc_write_coef_idx(codec, 0x43, 0x9614); } -- cgit From 307229d2ac5f60447cc655ebbce5d9c0baa18bbc Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Thu, 24 Oct 2013 21:10:34 +0300 Subject: ALSA: hda - hdmi: Allow HDA patches to customize more operations Upcoming AMD multichannel support requires many customized operations (channel mapping, ELD, HBR) but can otherwise share most of its code with the generic patch. Add a local struct hdmi_ops containing customizable HDMI-specific callbacks and move the current code to those callbacks. Functionality is unaltered. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 275 ++++++++++++++++++++++++++++++++------------- 1 file changed, 195 insertions(+), 80 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0f9b10322b2a..acc00fe01b71 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -82,6 +82,39 @@ struct hdmi_spec_per_pin { #endif }; +struct cea_channel_speaker_allocation; + +/* operations used by generic code that can be overridden by patches */ +struct hdmi_ops { + int (*pin_get_eld)(struct hda_codec *codec, hda_nid_t pin_nid, + unsigned char *buf, int *eld_size); + + /* get and set channel assigned to each HDMI ASP (audio sample packet) slot */ + int (*pin_get_slot_channel)(struct hda_codec *codec, hda_nid_t pin_nid, + int asp_slot); + int (*pin_set_slot_channel)(struct hda_codec *codec, hda_nid_t pin_nid, + int asp_slot, int channel); + + void (*pin_setup_infoframe)(struct hda_codec *codec, hda_nid_t pin_nid, + int ca, int active_channels, int conn_type); + + /* enable/disable HBR (HD passthrough) */ + int (*pin_hbr_setup)(struct hda_codec *codec, hda_nid_t pin_nid, bool hbr); + + int (*setup_stream)(struct hda_codec *codec, hda_nid_t cvt_nid, + hda_nid_t pin_nid, u32 stream_tag, int format); + + /* Helpers for producing the channel map TLVs. These can be overridden + * for devices that have non-standard mapping requirements. */ + int (*chmap_cea_alloc_validate_get_type)(struct cea_channel_speaker_allocation *cap, + int channels); + void (*cea_alloc_to_tlv_chmap)(struct cea_channel_speaker_allocation *cap, + unsigned int *chmap, int channels); + + /* check that the user-given chmap is supported */ + int (*chmap_validate)(int ca, int channels, unsigned char *chmap); +}; + struct hdmi_spec { int num_cvts; struct snd_array cvts; /* struct hdmi_spec_per_cvt */ @@ -93,6 +126,7 @@ struct hdmi_spec { unsigned int channels_max; /* max over all cvts */ struct hdmi_eld temp_eld; + struct hdmi_ops ops; /* * Non-generic ATI/NVIDIA specific */ @@ -648,24 +682,24 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid) { #ifdef CONFIG_SND_DEBUG_VERBOSE + struct hdmi_spec *spec = codec->spec; int i; - int slot; + int channel; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_CHAN_SLOT, i); + channel = spec->ops.pin_get_slot_channel(codec, pin_nid, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", - slot >> 4, slot & 0xf); + channel, i); } #endif } - static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, bool non_pcm, int ca) { + struct hdmi_spec *spec = codec->spec; struct cea_channel_speaker_allocation *ch_alloc; int i; int err; @@ -698,9 +732,10 @@ static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, } for (i = 0; i < 8; i++) { - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - non_pcm ? non_pcm_mapping[i] : hdmi_channel_mapping[ca][i]); + int slotsetup = non_pcm ? non_pcm_mapping[i] : hdmi_channel_mapping[ca][i]; + int hdmi_slot = slotsetup & 0x0f; + int channel = (slotsetup & 0xf0) >> 4; + err = spec->ops.pin_set_slot_channel(codec, pin_nid, hdmi_slot, channel); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -810,6 +845,7 @@ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, int chs, unsigned char *map, int ca) { + struct hdmi_spec *spec = codec->spec; int ordered_ca = get_channel_allocation_order(ca); int alsa_pos, hdmi_slot; int assignments[8] = {[0 ... 7] = 0xf}; @@ -825,11 +861,10 @@ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, } for (hdmi_slot = 0; hdmi_slot < 8; hdmi_slot++) { - int val, err; + int err; - val = (assignments[hdmi_slot] << 4) | hdmi_slot; - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, val); + err = spec->ops.pin_set_slot_channel(codec, pin_nid, hdmi_slot, + assignments[hdmi_slot]); if (err) return -EINVAL; } @@ -865,6 +900,22 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, hdmi_debug_channel_mapping(codec, pin_nid); } +static int hdmi_pin_set_slot_channel(struct hda_codec *codec, hda_nid_t pin_nid, + int asp_slot, int channel) +{ + return snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + (channel << 4) | asp_slot); +} + +static int hdmi_pin_get_slot_channel(struct hda_codec *codec, hda_nid_t pin_nid, + int asp_slot) +{ + return (snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_CHAN_SLOT, + asp_slot) & 0xf0) >> 4; +} + /* * Audio InfoFrame routines */ @@ -986,16 +1037,64 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } +static void hdmi_pin_setup_infoframe(struct hda_codec *codec, + hda_nid_t pin_nid, + int ca, int active_channels, + int conn_type) +{ + union audio_infoframe ai; + + if (conn_type == 0) { /* HDMI */ + struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; + + hdmi_ai->type = 0x84; + hdmi_ai->ver = 0x01; + hdmi_ai->len = 0x0a; + hdmi_ai->CC02_CT47 = active_channels - 1; + hdmi_ai->CA = ca; + hdmi_checksum_audio_infoframe(hdmi_ai); + } else if (conn_type == 1) { /* DisplayPort */ + struct dp_audio_infoframe *dp_ai = &ai.dp; + + dp_ai->type = 0x84; + dp_ai->len = 0x1b; + dp_ai->ver = 0x11 << 2; + dp_ai->CC02_CT47 = active_channels - 1; + dp_ai->CA = ca; + } else { + snd_printd("HDMI: unknown connection type at pin %d\n", + pin_nid); + return; + } + + /* + * sizeof(ai) is used instead of sizeof(*hdmi_ai) or + * sizeof(*dp_ai) to avoid partial match/update problems when + * the user switches between HDMI/DP monitors. + */ + if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, + sizeof(ai))) { + snd_printdd("hdmi_pin_setup_infoframe: " + "pin=%d channels=%d ca=0x%02x\n", + pin_nid, + active_channels, ca); + hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_fill_audio_infoframe(codec, pin_nid, + ai.bytes, sizeof(ai)); + hdmi_start_infoframe_trans(codec, pin_nid); + } +} + static void hdmi_setup_audio_infoframe(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, bool non_pcm) { + struct hdmi_spec *spec = codec->spec; hda_nid_t pin_nid = per_pin->pin_nid; int channels = per_pin->channels; int active_channels; struct hdmi_eld *eld; int ca, ordered_ca; - union audio_infoframe ai; if (!channels) return; @@ -1021,30 +1120,6 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hdmi_set_channel_count(codec, per_pin->cvt_nid, active_channels); - memset(&ai, 0, sizeof(ai)); - if (eld->info.conn_type == 0) { /* HDMI */ - struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; - - hdmi_ai->type = 0x84; - hdmi_ai->ver = 0x01; - hdmi_ai->len = 0x0a; - hdmi_ai->CC02_CT47 = active_channels - 1; - hdmi_ai->CA = ca; - hdmi_checksum_audio_infoframe(hdmi_ai); - } else if (eld->info.conn_type == 1) { /* DisplayPort */ - struct dp_audio_infoframe *dp_ai = &ai.dp; - - dp_ai->type = 0x84; - dp_ai->len = 0x1b; - dp_ai->ver = 0x11 << 2; - dp_ai->CC02_CT47 = active_channels - 1; - dp_ai->CA = ca; - } else { - snd_printd("HDMI: unknown connection type at pin %d\n", - pin_nid); - return; - } - /* * always configure channel mapping, it may have been changed by the * user in the meantime @@ -1053,27 +1128,12 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, channels, per_pin->chmap, per_pin->chmap_set); - /* - * sizeof(ai) is used instead of sizeof(*hdmi_ai) or - * sizeof(*dp_ai) to avoid partial match/update problems when - * the user switches between HDMI/DP monitors. - */ - if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, - sizeof(ai))) { - snd_printdd("hdmi_setup_audio_infoframe: " - "pin=%d channels=%d ca=0x%02x\n", - pin_nid, - active_channels, ca); - hdmi_stop_infoframe_trans(codec, pin_nid); - hdmi_fill_audio_infoframe(codec, pin_nid, - ai.bytes, sizeof(ai)); - hdmi_start_infoframe_trans(codec, pin_nid); - } + spec->ops.pin_setup_infoframe(codec, pin_nid, ca, active_channels, + eld->info.conn_type); per_pin->non_pcm = non_pcm; } - /* * Unsolicited events */ @@ -1176,26 +1236,22 @@ static void haswell_verify_D0(struct hda_codec *codec, #define is_hbr_format(format) \ ((format & AC_FMT_TYPE_NON_PCM) && (format & AC_FMT_CHAN_MASK) == 7) -static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, - hda_nid_t pin_nid, u32 stream_tag, int format) +static int hdmi_pin_hbr_setup(struct hda_codec *codec, hda_nid_t pin_nid, + bool hbr) { - int pinctl; - int new_pinctl = 0; - - if (is_haswell(codec)) - haswell_verify_D0(codec, cvt_nid, pin_nid); + int pinctl, new_pinctl; if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); new_pinctl = pinctl & ~AC_PINCTL_EPT; - if (is_hbr_format(format)) + if (hbr) new_pinctl |= AC_PINCTL_EPT_HBR; else new_pinctl |= AC_PINCTL_EPT_NATIVE; - snd_printdd("hdmi_setup_stream: " + snd_printdd("hdmi_pin_hbr_setup: " "NID=0x%x, %spinctl=0x%x\n", pin_nid, pinctl == new_pinctl ? "" : "new-", @@ -1205,11 +1261,26 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, new_pinctl); + } else if (hbr) + return -EINVAL; - } - if (is_hbr_format(format) && !new_pinctl) { + return 0; +} + +static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, + hda_nid_t pin_nid, u32 stream_tag, int format) +{ + struct hdmi_spec *spec = codec->spec; + int err; + + if (is_haswell(codec)) + haswell_verify_D0(codec, cvt_nid, pin_nid); + + err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); + + if (err) { snd_printdd("hdmi_setup_stream: HBR is not supported\n"); - return -EINVAL; + return err; } snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); @@ -1424,7 +1495,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { - if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, + if (spec->ops.pin_get_eld(codec, pin_nid, eld->eld_buffer, &eld->eld_size) < 0) eld->eld_valid = false; else { @@ -1654,7 +1725,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); mutex_unlock(&per_pin->lock); - return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); + return spec->ops.setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -1726,6 +1797,34 @@ static int hdmi_chmap_ctl_info(struct snd_kcontrol *kcontrol, return 0; } +static int hdmi_chmap_cea_alloc_validate_get_type(struct cea_channel_speaker_allocation *cap, + int channels) +{ + /* If the speaker allocation matches the channel count, it is OK.*/ + if (cap->channels != channels) + return -1; + + /* all channels are remappable freely */ + return SNDRV_CTL_TLVT_CHMAP_VAR; +} + +static void hdmi_cea_alloc_to_tlv_chmap(struct cea_channel_speaker_allocation *cap, + unsigned int *chmap, int channels) +{ + int count = 0; + int c; + + for (c = 7; c >= 0; c--) { + int spk = cap->speakers[c]; + if (!spk) + continue; + + chmap[count++] = spk_to_chmap(spk); + } + + WARN_ON(count != channels); +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { @@ -1742,16 +1841,19 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, size -= 8; dst = tlv + 2; for (chs = 2; chs <= spec->channels_max; chs++) { - int i, c; + int i; struct cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - if (cap->channels != chs) + int type = spec->ops.chmap_cea_alloc_validate_get_type(cap, chs); + unsigned int tlv_chmap[8]; + + if (type < 0) continue; if (size < 8) return -ENOMEM; - if (put_user(SNDRV_CTL_TLVT_CHMAP_VAR, dst) || + if (put_user(type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; dst += 2; @@ -1761,14 +1863,10 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -ENOMEM; size -= chs_bytes; count += chs_bytes; - for (c = 7; c >= 0; c--) { - int spk = cap->speakers[c]; - if (!spk) - continue; - if (put_user(spk_to_chmap(spk), dst)) - return -EFAULT; - dst++; - } + spec->ops.cea_alloc_to_tlv_chmap(cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) + return -EFAULT; + dst += chs; } } if (put_user(count, tlv + 1)) @@ -1802,7 +1900,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, unsigned int ctl_idx; struct snd_pcm_substream *substream; unsigned char chmap[8]; - int i, ca, prepared = 0; + int i, err, ca, prepared = 0; ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); substream = snd_pcm_chmap_substream(info, ctl_idx); @@ -1826,6 +1924,11 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); if (ca < 0) return -EINVAL; + if (spec->ops.chmap_validate) { + err = spec->ops.chmap_validate(ca, ARRAY_SIZE(chmap), chmap); + if (err) + return err; + } mutex_lock(&per_pin->lock); per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); @@ -2032,6 +2135,17 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { #endif }; +static const struct hdmi_ops generic_standard_hdmi_ops = { + .pin_get_eld = snd_hdmi_get_eld, + .pin_get_slot_channel = hdmi_pin_get_slot_channel, + .pin_set_slot_channel = hdmi_pin_set_slot_channel, + .pin_setup_infoframe = hdmi_pin_setup_infoframe, + .pin_hbr_setup = hdmi_pin_hbr_setup, + .setup_stream = hdmi_setup_stream, + .chmap_cea_alloc_validate_get_type = hdmi_chmap_cea_alloc_validate_get_type, + .cea_alloc_to_tlv_chmap = hdmi_cea_alloc_to_tlv_chmap, +}; + static void intel_haswell_fixup_connect_list(struct hda_codec *codec, hda_nid_t nid) @@ -2114,6 +2228,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + spec->ops = generic_standard_hdmi_ops; codec->spec = spec; hdmi_array_init(spec, 4); -- cgit From 5a61358433b1f89b500f2c365746c73cb7a27e2f Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Thu, 24 Oct 2013 21:10:35 +0300 Subject: ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ATI/AMD codecs do not support all the standard HDA HDMI/DP functions, instead various vendor-specific verbs are provided. This commit addresses these missing functions: - standard channel mapping support - standard infoframe configuration support ATI/AMD provides their own verbs that allow the following: - setting CA for infoframe - setting down-mix information for infoframe - channel pair remapping - individual channel remapping (revision ID 3+, 0x100300+) The documentation for the verbs has now been released by AMD: http://www.x.org/docs/AMD/AMD_HDA_verbs_v2.pdf Add support for the ATI/AMD specific verbs and use them instead of the generic methods on ATI/AMD codecs. This allows multi-channel PCM audio to work. Channel remapping is restricted to pairwise mapping on codecs with revision ID 2 (0x100200 as reported by procfs codec#X) or lower. This means cards up to Radeon HD7670 as far as I know. This will not affect standard multi-channel modes since these codecs support automatic FC-LFE swapping for HDMI. ATI/AMD codecs do not advertise all of their supported rates, formats and channel counts, therefore that information is forced accordingly so that all HDMI 1.x PCM parameters are marked as supported. Support for multiple ports is also added to patch_atihdmi so that 0x1002aa01 codecs with multiple ports will work properly when switched back to that patch. v2: splitted ELD emulation to a separate patch, tlv fixes v3: adapted to the new hdmi_ops infrastructure, fixed rev3+ vendor id Signed-off-by: Anssi Hannula Tested-by: Peter Frühberger # v2 Tested-by: Olivier Langlois # v2+rev3fix Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 302 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 273 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index acc00fe01b71..1114be018e5a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -6,6 +6,7 @@ * Copyright (c) 2006 ATI Technologies Inc. * Copyright (c) 2008 NVIDIA Corp. All rights reserved. * Copyright (c) 2008 Wei Ni + * Copyright (c) 2013 Anssi Hannula * * Authors: * Wu Fengguang @@ -128,7 +129,7 @@ struct hdmi_spec { struct hdmi_eld temp_eld; struct hdmi_ops ops; /* - * Non-generic ATI/NVIDIA specific + * Non-generic VIA/NVIDIA specific */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; @@ -2784,49 +2785,292 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) } /* - * ATI-specific implementations - * - * FIXME: we may omit the whole this and use the generic code once after - * it's confirmed to work. + * ATI/AMD-specific implementations */ -#define ATIHDMI_CVT_NID 0x02 /* audio converter */ -#define ATIHDMI_PIN_NID 0x03 /* HDMI output pin */ +#define is_amdhdmi_rev3_or_later(codec) \ + ((codec)->vendor_id == 0x1002aa01 && ((codec)->revision_id & 0xff00) >= 0x0300) +#define has_amd_full_remap_support(codec) is_amdhdmi_rev3_or_later(codec) + +/* ATI/AMD specific HDA pin verbs, see the AMD HDA Verbs specification */ +#define ATI_VERB_SET_CHANNEL_ALLOCATION 0x771 +#define ATI_VERB_SET_DOWNMIX_INFO 0x772 +#define ATI_VERB_SET_MULTICHANNEL_01 0x777 +#define ATI_VERB_SET_MULTICHANNEL_23 0x778 +#define ATI_VERB_SET_MULTICHANNEL_45 0x779 +#define ATI_VERB_SET_MULTICHANNEL_67 0x77a +#define ATI_VERB_SET_MULTICHANNEL_1 0x785 +#define ATI_VERB_SET_MULTICHANNEL_3 0x786 +#define ATI_VERB_SET_MULTICHANNEL_5 0x787 +#define ATI_VERB_SET_MULTICHANNEL_7 0x788 +#define ATI_VERB_SET_MULTICHANNEL_MODE 0x789 +#define ATI_VERB_GET_CHANNEL_ALLOCATION 0xf71 +#define ATI_VERB_GET_DOWNMIX_INFO 0xf72 +#define ATI_VERB_GET_MULTICHANNEL_01 0xf77 +#define ATI_VERB_GET_MULTICHANNEL_23 0xf78 +#define ATI_VERB_GET_MULTICHANNEL_45 0xf79 +#define ATI_VERB_GET_MULTICHANNEL_67 0xf7a +#define ATI_VERB_GET_MULTICHANNEL_1 0xf85 +#define ATI_VERB_GET_MULTICHANNEL_3 0xf86 +#define ATI_VERB_GET_MULTICHANNEL_5 0xf87 +#define ATI_VERB_GET_MULTICHANNEL_7 0xf88 +#define ATI_VERB_GET_MULTICHANNEL_MODE 0xf89 + +#define ATI_OUT_ENABLE 0x1 + +#define ATI_MULTICHANNEL_MODE_PAIRED 0 +#define ATI_MULTICHANNEL_MODE_SINGLE 1 + +static void atihdmi_pin_setup_infoframe(struct hda_codec *codec, hda_nid_t pin_nid, int ca, + int active_channels, int conn_type) +{ + snd_hda_codec_write(codec, pin_nid, 0, ATI_VERB_SET_CHANNEL_ALLOCATION, ca); +} + +static int atihdmi_paired_swap_fc_lfe(int pos) +{ + /* + * ATI/AMD have automatic FC/LFE swap built-in + * when in pairwise mapping mode. + */ + + switch (pos) { + /* see channel_allocations[].speakers[] */ + case 2: return 3; + case 3: return 2; + default: break; + } + + return pos; +} + +static int atihdmi_paired_chmap_validate(int ca, int chs, unsigned char *map) +{ + struct cea_channel_speaker_allocation *cap; + int i, j; + + /* check that only channel pairs need to be remapped on old pre-rev3 ATI/AMD */ + + cap = &channel_allocations[get_channel_allocation_order(ca)]; + for (i = 0; i < chs; ++i) { + int mask = to_spk_mask(map[i]); + bool ok = false; + bool companion_ok = false; + + if (!mask) + continue; + + for (j = 0 + i % 2; j < 8; j += 2) { + int chan_idx = 7 - atihdmi_paired_swap_fc_lfe(j); + if (cap->speakers[chan_idx] == mask) { + /* channel is in a supported position */ + ok = true; + + if (i % 2 == 0 && i + 1 < chs) { + /* even channel, check the odd companion */ + int comp_chan_idx = 7 - atihdmi_paired_swap_fc_lfe(j + 1); + int comp_mask_req = to_spk_mask(map[i+1]); + int comp_mask_act = cap->speakers[comp_chan_idx]; + + if (comp_mask_req == comp_mask_act) + companion_ok = true; + else + return -EINVAL; + } + break; + } + } + + if (!ok) + return -EINVAL; + + if (companion_ok) + i++; /* companion channel already checked */ + } + + return 0; +} + +static int atihdmi_pin_set_slot_channel(struct hda_codec *codec, hda_nid_t pin_nid, + int hdmi_slot, int stream_channel) +{ + int verb; + int ati_channel_setup = 0; + + if (hdmi_slot > 7) + return -EINVAL; + + if (!has_amd_full_remap_support(codec)) { + hdmi_slot = atihdmi_paired_swap_fc_lfe(hdmi_slot); + + /* In case this is an odd slot but without stream channel, do not + * disable the slot since the corresponding even slot could have a + * channel. In case neither have a channel, the slot pair will be + * disabled when this function is called for the even slot. */ + if (hdmi_slot % 2 != 0 && stream_channel == 0xf) + return 0; + + hdmi_slot -= hdmi_slot % 2; + + if (stream_channel != 0xf) + stream_channel -= stream_channel % 2; + } + + verb = ATI_VERB_SET_MULTICHANNEL_01 + hdmi_slot/2 + (hdmi_slot % 2) * 0x00e; + + /* ati_channel_setup format: [7..4] = stream_channel_id, [1] = mute, [0] = enable */ + + if (stream_channel != 0xf) + ati_channel_setup = (stream_channel << 4) | ATI_OUT_ENABLE; + + return snd_hda_codec_write(codec, pin_nid, 0, verb, ati_channel_setup); +} + +static int atihdmi_pin_get_slot_channel(struct hda_codec *codec, hda_nid_t pin_nid, + int asp_slot) +{ + bool was_odd = false; + int ati_asp_slot = asp_slot; + int verb; + int ati_channel_setup; + + if (asp_slot > 7) + return -EINVAL; + + if (!has_amd_full_remap_support(codec)) { + ati_asp_slot = atihdmi_paired_swap_fc_lfe(asp_slot); + if (ati_asp_slot % 2 != 0) { + ati_asp_slot -= 1; + was_odd = true; + } + } + + verb = ATI_VERB_GET_MULTICHANNEL_01 + ati_asp_slot/2 + (ati_asp_slot % 2) * 0x00e; + + ati_channel_setup = snd_hda_codec_read(codec, pin_nid, 0, verb, 0); + + if (!(ati_channel_setup & ATI_OUT_ENABLE)) + return 0xf; + + return ((ati_channel_setup & 0xf0) >> 4) + !!was_odd; +} -static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int atihdmi_paired_chmap_cea_alloc_validate_get_type(struct cea_channel_speaker_allocation *cap, + int channels) +{ + int c; + + /* + * Pre-rev3 ATI/AMD codecs operate in a paired channel mode, so + * we need to take that into account (a single channel may take 2 + * channel slots if we need to carry a silent channel next to it). + * On Rev3+ AMD codecs this function is not used. + */ + int chanpairs = 0; + + /* We only produce even-numbered channel count TLVs */ + if ((channels % 2) != 0) + return -1; + + for (c = 0; c < 7; c += 2) { + if (cap->speakers[c] || cap->speakers[c+1]) + chanpairs++; + } + + if (chanpairs * 2 != channels) + return -1; + + return SNDRV_CTL_TLVT_CHMAP_PAIRED; +} + +static void atihdmi_paired_cea_alloc_to_tlv_chmap(struct cea_channel_speaker_allocation *cap, + unsigned int *chmap, int channels) +{ + /* produce paired maps for pre-rev3 ATI/AMD codecs */ + int count = 0; + int c; + + for (c = 7; c >= 0; c--) { + int chan = 7 - atihdmi_paired_swap_fc_lfe(7 - c); + int spk = cap->speakers[chan]; + if (!spk) { + /* add N/A channel if the companion channel is occupied */ + if (cap->speakers[chan + (chan % 2 ? -1 : 1)]) + chmap[count++] = SNDRV_CHMAP_NA; + + continue; + } + + chmap[count++] = spk_to_chmap(spk); + } + + WARN_ON(count != channels); +} + +static int atihdmi_init(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_cvt *per_cvt = get_cvt(spec, 0); - int chans = substream->runtime->channels; - int i, err; + int pin_idx, err; - err = simple_playback_pcm_prepare(hinfo, codec, stream_tag, format, - substream); - if (err < 0) + err = generic_hdmi_init(codec); + + if (err) return err; - snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chans - 1); - /* FIXME: XXX */ - for (i = 0; i < chans; i++) { - snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - (i << 4) | i); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + + /* make sure downmix information in infoframe is zero */ + snd_hda_codec_write(codec, per_pin->pin_nid, 0, ATI_VERB_SET_DOWNMIX_INFO, 0); + + /* enable channel-wise remap mode if supported */ + if (has_amd_full_remap_support(codec)) + snd_hda_codec_write(codec, per_pin->pin_nid, 0, + ATI_VERB_SET_MULTICHANNEL_MODE, + ATI_MULTICHANNEL_MODE_SINGLE); } + return 0; } static int patch_atihdmi(struct hda_codec *codec) { struct hdmi_spec *spec; - int err = patch_simple_hdmi(codec, ATIHDMI_CVT_NID, ATIHDMI_PIN_NID); - if (err < 0) + struct hdmi_spec_per_cvt *per_cvt; + int err, cvt_idx; + + err = patch_generic_hdmi(codec); + + if (err) return err; + + codec->patch_ops.init = atihdmi_init; + spec = codec->spec; - spec->pcm_playback.ops.prepare = atihdmi_playback_pcm_prepare; + + spec->ops.pin_get_slot_channel = atihdmi_pin_get_slot_channel; + spec->ops.pin_set_slot_channel = atihdmi_pin_set_slot_channel; + spec->ops.pin_setup_infoframe = atihdmi_pin_setup_infoframe; + + if (!has_amd_full_remap_support(codec)) { + /* override to ATI/AMD-specific versions with pairwise mapping */ + spec->ops.chmap_cea_alloc_validate_get_type = + atihdmi_paired_chmap_cea_alloc_validate_get_type; + spec->ops.cea_alloc_to_tlv_chmap = atihdmi_paired_cea_alloc_to_tlv_chmap; + spec->ops.chmap_validate = atihdmi_paired_chmap_validate; + } + + /* ATI/AMD converters do not advertise all of their capabilities */ + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); + per_cvt->channels_max = max(per_cvt->channels_max, 8u); + per_cvt->rates |= SUPPORTED_RATES; + per_cvt->formats |= SUPPORTED_FORMATS; + per_cvt->maxbps = max(per_cvt->maxbps, 24u); + } + + spec->channels_max = max(spec->channels_max, 8u); + return 0; } @@ -2846,7 +3090,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, -{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi }, -- cgit From 89250f84644676c1ed6a09acef1033e14596a402 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Thu, 24 Oct 2013 21:10:36 +0300 Subject: ALSA: hda - hdmi: Add ELD emulation for ATI/AMD codecs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ATI/AMD HDMI/DP codecs do not include standard HDA ELD (EDID-like data) support. In place of providing access to an ELD buffer, various vendor-specific verbs are provided to provide the relevant information. Revision ID 3 and later (0x100300 as reported by procfs codec#X) have support for providing more information than the previous revisions (but only if supported by the display driver). Generate ELD from the information provided by the vendor-specific verbs on ATI/AMD codecs. The specification is available at: http://www.x.org/docs/AMD/AMD_HDA_verbs_v2.pdf v2: moved code to hda_eld.c and cleaned it up v3: adapted to hdmi_ops infrastructure Signed-off-by: Anssi Hannula Tested-by: Peter Frühberger # v2 Tested-by: Olivier Langlois # v2 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 151 +++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_local.h | 4 ++ sound/pci/hda/patch_hdmi.c | 9 +++ 3 files changed, 164 insertions(+) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index f62356c2f54c..32d3e3855a6e 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -2,6 +2,7 @@ * Generic routines and proc interface for ELD(EDID Like Data) information * * Copyright(c) 2008 Intel Corporation. + * Copyright (c) 2013 Anssi Hannula * * Authors: * Wu Fengguang @@ -633,3 +634,153 @@ void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, hinfo->maxbps = min(hinfo->maxbps, maxbps); hinfo->channels_max = min(hinfo->channels_max, channels_max); } + + +/* ATI/AMD specific stuff (ELD emulation) */ + +#define ATI_VERB_SET_AUDIO_DESCRIPTOR 0x776 +#define ATI_VERB_SET_SINK_INFO_INDEX 0x780 +#define ATI_VERB_GET_SPEAKER_ALLOCATION 0xf70 +#define ATI_VERB_GET_AUDIO_DESCRIPTOR 0xf76 +#define ATI_VERB_GET_AUDIO_VIDEO_DELAY 0xf7b +#define ATI_VERB_GET_SINK_INFO_INDEX 0xf80 +#define ATI_VERB_GET_SINK_INFO_DATA 0xf81 + +#define ATI_SPKALLOC_SPKALLOC 0x007f +#define ATI_SPKALLOC_TYPE_HDMI 0x0100 +#define ATI_SPKALLOC_TYPE_DISPLAYPORT 0x0200 + +/* first three bytes are just standard SAD */ +#define ATI_AUDIODESC_CHANNELS 0x00000007 +#define ATI_AUDIODESC_RATES 0x0000ff00 +#define ATI_AUDIODESC_LPCM_STEREO_RATES 0xff000000 + +/* in standard HDMI VSDB format */ +#define ATI_DELAY_VIDEO_LATENCY 0x000000ff +#define ATI_DELAY_AUDIO_LATENCY 0x0000ff00 + +enum ati_sink_info_idx { + ATI_INFO_IDX_MANUFACTURER_ID = 0, + ATI_INFO_IDX_PRODUCT_ID = 1, + ATI_INFO_IDX_SINK_DESC_LEN = 2, + ATI_INFO_IDX_PORT_ID_LOW = 3, + ATI_INFO_IDX_PORT_ID_HIGH = 4, + ATI_INFO_IDX_SINK_DESC_FIRST = 5, + ATI_INFO_IDX_SINK_DESC_LAST = 22, /* max len 18 bytes */ +}; + +int snd_hdmi_get_eld_ati(struct hda_codec *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size, bool rev3_or_later) +{ + int spkalloc, ati_sad, aud_synch; + int sink_desc_len = 0; + int pos, i; + + /* ATI/AMD does not have ELD, emulate it */ + + spkalloc = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SPEAKER_ALLOCATION, 0); + + if (!spkalloc) { + snd_printd(KERN_INFO "HDMI ATI/AMD: no speaker allocation for ELD\n"); + return -EINVAL; + } + + memset(buf, 0, ELD_FIXED_BYTES + ELD_MAX_MNL + ELD_MAX_SAD * 3); + + /* version */ + buf[0] = ELD_VER_CEA_861D << 3; + + /* speaker allocation from EDID */ + buf[7] = spkalloc & ATI_SPKALLOC_SPKALLOC; + + /* is DisplayPort? */ + if (spkalloc & ATI_SPKALLOC_TYPE_DISPLAYPORT) + buf[5] |= 0x04; + + pos = ELD_FIXED_BYTES; + + if (rev3_or_later) { + int sink_info; + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_SINK_INFO_INDEX, ATI_INFO_IDX_PORT_ID_LOW); + sink_info = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SINK_INFO_DATA, 0); + put_unaligned_le32(sink_info, buf + 8); + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_SINK_INFO_INDEX, ATI_INFO_IDX_PORT_ID_HIGH); + sink_info = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SINK_INFO_DATA, 0); + put_unaligned_le32(sink_info, buf + 12); + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_SINK_INFO_INDEX, ATI_INFO_IDX_MANUFACTURER_ID); + sink_info = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SINK_INFO_DATA, 0); + put_unaligned_le16(sink_info, buf + 16); + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_SINK_INFO_INDEX, ATI_INFO_IDX_PRODUCT_ID); + sink_info = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SINK_INFO_DATA, 0); + put_unaligned_le16(sink_info, buf + 18); + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_SINK_INFO_INDEX, ATI_INFO_IDX_SINK_DESC_LEN); + sink_desc_len = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SINK_INFO_DATA, 0); + + if (sink_desc_len > ELD_MAX_MNL) { + snd_printd(KERN_INFO "HDMI ATI/AMD: Truncating HDMI sink description with length %d\n", + sink_desc_len); + sink_desc_len = ELD_MAX_MNL; + } + + buf[4] |= sink_desc_len; + + for (i = 0; i < sink_desc_len; i++) { + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_SINK_INFO_INDEX, ATI_INFO_IDX_SINK_DESC_FIRST + i); + buf[pos++] = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SINK_INFO_DATA, 0); + } + } + + for (i = AUDIO_CODING_TYPE_LPCM; i <= AUDIO_CODING_TYPE_WMAPRO; i++) { + if (i == AUDIO_CODING_TYPE_SACD || i == AUDIO_CODING_TYPE_DST) + continue; /* not handled by ATI/AMD */ + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_AUDIO_DESCRIPTOR, i << 3); + ati_sad = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_AUDIO_DESCRIPTOR, 0); + + if (ati_sad & ATI_AUDIODESC_RATES) { + /* format is supported, copy SAD as-is */ + buf[pos++] = (ati_sad & 0x0000ff) >> 0; + buf[pos++] = (ati_sad & 0x00ff00) >> 8; + buf[pos++] = (ati_sad & 0xff0000) >> 16; + } + + if (i == AUDIO_CODING_TYPE_LPCM + && (ati_sad & ATI_AUDIODESC_LPCM_STEREO_RATES) + && (ati_sad & ATI_AUDIODESC_LPCM_STEREO_RATES) >> 16 != (ati_sad & ATI_AUDIODESC_RATES)) { + /* for PCM there is a separate stereo rate mask */ + buf[pos++] = ((ati_sad & 0x000000ff) & ~ATI_AUDIODESC_CHANNELS) | 0x1; + /* rates from the extra byte */ + buf[pos++] = (ati_sad & 0xff000000) >> 24; + buf[pos++] = (ati_sad & 0x00ff0000) >> 16; + } + } + + if (pos == ELD_FIXED_BYTES + sink_desc_len) { + snd_printd(KERN_INFO "HDMI ATI/AMD: no audio descriptors for ELD\n"); + return -EINVAL; + } + + aud_synch = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_AUDIO_VIDEO_DELAY, 0); + if ((aud_synch & ATI_DELAY_VIDEO_LATENCY) && (aud_synch & ATI_DELAY_AUDIO_LATENCY)) { + int video_latency = (aud_synch & ATI_DELAY_VIDEO_LATENCY) - 1; + int audio_latency = ((aud_synch & ATI_DELAY_AUDIO_LATENCY) >> 8) - 1; + + if (video_latency > audio_latency) + buf[6] = min(video_latency - audio_latency, 0xfa); + } + + /* Baseline length */ + buf[2] = pos - 4; + + /* SAD count */ + buf[5] |= ((pos - ELD_FIXED_BYTES - sink_desc_len) / 3) << 4; + + *eld_size = pos; + + return 0; +} diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 46cddd4c7b72..d398b648bb5d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -763,6 +763,10 @@ void snd_hdmi_show_eld(struct parsed_hdmi_eld *e); void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, struct hda_pcm_stream *hinfo); +int snd_hdmi_get_eld_ati(struct hda_codec *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size, + bool rev3_or_later); + #ifdef CONFIG_PROC_FS void snd_hdmi_print_eld_info(struct hdmi_eld *eld, struct snd_info_buffer *buffer); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1114be018e5a..0307ae87bef3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2821,6 +2821,14 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) #define ATI_MULTICHANNEL_MODE_PAIRED 0 #define ATI_MULTICHANNEL_MODE_SINGLE 1 +static int atihdmi_pin_get_eld(struct hda_codec *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size) +{ + /* call hda_eld.c ATI/AMD-specific function */ + return snd_hdmi_get_eld_ati(codec, nid, buf, eld_size, + is_amdhdmi_rev3_or_later(codec)); +} + static void atihdmi_pin_setup_infoframe(struct hda_codec *codec, hda_nid_t pin_nid, int ca, int active_channels, int conn_type) { @@ -3048,6 +3056,7 @@ static int patch_atihdmi(struct hda_codec *codec) spec = codec->spec; + spec->ops.pin_get_eld = atihdmi_pin_get_eld; spec->ops.pin_get_slot_channel = atihdmi_pin_get_slot_channel; spec->ops.pin_set_slot_channel = atihdmi_pin_set_slot_channel; spec->ops.pin_setup_infoframe = atihdmi_pin_setup_infoframe; -- cgit From 461cf6b39dded3ddb15a2300a534aba039870e5f Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Thu, 24 Oct 2013 21:10:37 +0300 Subject: ALSA: hda - hdmi: Add HBR bitstreaming support for ATI/AMD HDMI codecs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ATI/AMD HDMI codecs do not include standard HDA HDMI HBR support (which is required for bitstreaming DTS-HD and Dolby TrueHD), instead they have custom verbs for checking and enabling it. Add support for the ATI/AMD HDMI HBR verbs. The specification is available at: http://www.x.org/docs/AMD/AMD_HDA_verbs_v2.pdf v2: adapted to hdmi_ops infrastructure Signed-off-by: Anssi Hannula Tested-by: Peter Frühberger # v1 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0307ae87bef3..f6a97607e522 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2799,6 +2799,7 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) #define ATI_VERB_SET_MULTICHANNEL_23 0x778 #define ATI_VERB_SET_MULTICHANNEL_45 0x779 #define ATI_VERB_SET_MULTICHANNEL_67 0x77a +#define ATI_VERB_SET_HBR_CONTROL 0x77c #define ATI_VERB_SET_MULTICHANNEL_1 0x785 #define ATI_VERB_SET_MULTICHANNEL_3 0x786 #define ATI_VERB_SET_MULTICHANNEL_5 0x787 @@ -2810,6 +2811,7 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) #define ATI_VERB_GET_MULTICHANNEL_23 0xf78 #define ATI_VERB_GET_MULTICHANNEL_45 0xf79 #define ATI_VERB_GET_MULTICHANNEL_67 0xf7a +#define ATI_VERB_GET_HBR_CONTROL 0xf7c #define ATI_VERB_GET_MULTICHANNEL_1 0xf85 #define ATI_VERB_GET_MULTICHANNEL_3 0xf86 #define ATI_VERB_GET_MULTICHANNEL_5 0xf87 @@ -2821,6 +2823,9 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) #define ATI_MULTICHANNEL_MODE_PAIRED 0 #define ATI_MULTICHANNEL_MODE_SINGLE 1 +#define ATI_HBR_CAPABLE 0x01 +#define ATI_HBR_ENABLE 0x10 + static int atihdmi_pin_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size) { @@ -3015,6 +3020,35 @@ static void atihdmi_paired_cea_alloc_to_tlv_chmap(struct cea_channel_speaker_all WARN_ON(count != channels); } +static int atihdmi_pin_hbr_setup(struct hda_codec *codec, hda_nid_t pin_nid, + bool hbr) +{ + int hbr_ctl, hbr_ctl_new; + + hbr_ctl = snd_hda_codec_read(codec, pin_nid, 0, ATI_VERB_GET_HBR_CONTROL, 0); + if (hbr_ctl & ATI_HBR_CAPABLE) { + if (hbr) + hbr_ctl_new = hbr_ctl | ATI_HBR_ENABLE; + else + hbr_ctl_new = hbr_ctl & ~ATI_HBR_ENABLE; + + snd_printdd("atihdmi_pin_hbr_setup: " + "NID=0x%x, %shbr-ctl=0x%x\n", + pin_nid, + hbr_ctl == hbr_ctl_new ? "" : "new-", + hbr_ctl_new); + + if (hbr_ctl != hbr_ctl_new) + snd_hda_codec_write(codec, pin_nid, 0, + ATI_VERB_SET_HBR_CONTROL, + hbr_ctl_new); + + } else if (hbr) + return -EINVAL; + + return 0; +} + static int atihdmi_init(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -3060,6 +3094,7 @@ static int patch_atihdmi(struct hda_codec *codec) spec->ops.pin_get_slot_channel = atihdmi_pin_get_slot_channel; spec->ops.pin_set_slot_channel = atihdmi_pin_set_slot_channel; spec->ops.pin_setup_infoframe = atihdmi_pin_setup_infoframe; + spec->ops.pin_hbr_setup = atihdmi_pin_hbr_setup; if (!has_amd_full_remap_support(codec)) { /* override to ATI/AMD-specific versions with pairwise mapping */ -- cgit From 84d69e790f48ff2e8472a8cf602243e43683eaf0 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Thu, 24 Oct 2013 21:10:38 +0300 Subject: ALSA: hda - hdmi: Disable ramp-up/down for non-PCM on AMD codecs Recent AMD HDMI codecs (revision ID 3 and later, 0x100300 as reported by procfs codec#0) have a configurable ramp-up/down functionality. The documentation ( http://www.x.org/docs/AMD/AMD_HDA_verbs_v2.pdf ) specifies that 180 ("180/256 =~ 0.7") is recommended for PCM and 0 for non-PCM. Apply the recommended values according to provided S/PDIF AES0 settings since ramp-up/down does not make sense for non-PCM. v2: adapted to hdmi_ops infrastructure * More note from Anssi: actually, re-reading mails reveals that Olivier didn't find the expected difference with this setting, except for "maybe slightly slower startup with AES0=6" (i.e. value 0, which is unexpected). So maybe a) it makes too unnoticiable a difference, or b) only affects certain hardware (card and/or sink), or c) ramp-up/down is only triggered with the MUTE bit of ATI_VERB_SET_MULTICHANNEL_xx which is also rev3+ specific, but is not presently used by the driver, or something else. So there's a significant chance setting ramp rate is useless for us ATM, but probably does not do actual harm either. Signed-off-by: Anssi Hannula Tested-by: Olivier Langlois # v1 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f6a97607e522..7bd89bf62559 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2818,6 +2818,10 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) #define ATI_VERB_GET_MULTICHANNEL_7 0xf88 #define ATI_VERB_GET_MULTICHANNEL_MODE 0xf89 +/* AMD specific HDA cvt verbs */ +#define ATI_VERB_SET_RAMP_RATE 0x770 +#define ATI_VERB_GET_RAMP_RATE 0xf70 + #define ATI_OUT_ENABLE 0x1 #define ATI_MULTICHANNEL_MODE_PAIRED 0 @@ -3049,6 +3053,23 @@ static int atihdmi_pin_hbr_setup(struct hda_codec *codec, hda_nid_t pin_nid, return 0; } +static int atihdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, + hda_nid_t pin_nid, u32 stream_tag, int format) +{ + + if (is_amdhdmi_rev3_or_later(codec)) { + int ramp_rate = 180; /* default as per AMD spec */ + /* disable ramp-up/down for non-pcm as per AMD spec */ + if (format & AC_FMT_TYPE_NON_PCM) + ramp_rate = 0; + + snd_hda_codec_write(codec, cvt_nid, 0, ATI_VERB_SET_RAMP_RATE, ramp_rate); + } + + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); +} + + static int atihdmi_init(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -3095,6 +3116,7 @@ static int patch_atihdmi(struct hda_codec *codec) spec->ops.pin_set_slot_channel = atihdmi_pin_set_slot_channel; spec->ops.pin_setup_infoframe = atihdmi_pin_setup_infoframe; spec->ops.pin_hbr_setup = atihdmi_pin_hbr_setup; + spec->ops.setup_stream = atihdmi_setup_stream; if (!has_amd_full_remap_support(codec)) { /* override to ATI/AMD-specific versions with pairwise mapping */ -- cgit From 7342017f4a0f129d277f78b8761f2732661ba30a Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Fri, 25 Oct 2013 01:45:18 +0300 Subject: ALSA: hda - hdmi: Re-setup pin and infoframe on plug-in on all codecs hdmi_setup_audio_infoframe() does not set up pin and infoframe if there is no connected sink. If a sink is connected while audio playback is already in progress, the pin and infoframe will not be properly set up, causing no audio or wrongly mapped audio. On Intel Haswell codecs the hdmi_setup_audio_infoframe() is already called again from hdmi_present_sense() when an ELD appears because transcoder:port mapping may have changed. Make the call non-Haswell-specific so that audio will be properly set up if the playback was started before a sink was connected. Tested on non-Haswell Intel HDMI codec by plugging sink in during playback. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7bd89bf62559..772827b9fb20 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1534,11 +1534,12 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; - /* Haswell-specific workaround: re-setup when the transcoder is - * changed during the stream playback + /* + * Re-setup pin and infoframe. This is needed e.g. when + * - sink is first plugged-in (infoframe is not set up if !monitor_present) + * - transcoder can change during stream playback on Haswell */ - if (is_haswell(codec) && - eld->eld_valid && !old_eld_valid && per_pin->setup) + if (eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } -- cgit From 99d8d3ba512810feb13e158042975dcda75cef31 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Thu, 24 Oct 2013 17:55:18 +0200 Subject: ASoC: kirkwood: Fix compile error due to patch 'add S/PDIF support' This patch fixes the compilation error of kirkwood-i2s.c introduced by the commit 75b9b65ee5a80e99e 'ASoC: kirkwood: add S/PDIF support'. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9ec38d15df9e..d34d91743e3f 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -568,7 +568,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } -- cgit From 29ca9c73e54131c9ad90c5381f368d9b09b5aca4 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 25 Oct 2013 17:06:24 +0800 Subject: ASoC: samsung: fix return value check in i2s_alloc_dai() In case of error, the function platform_device_alloc() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 32956df8f50c..2e031fa729f0 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1073,7 +1073,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); - if (IS_ERR(i2s->pdev)) + if (!i2s->pdev) return NULL; i2s->pdev->dev.parent = &pdev->dev; -- cgit From 6dd17757927ba9d23c604fee6fe72b4755c7ea7f Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 25 Oct 2013 10:01:14 -0500 Subject: ASoC: cs42l52: Add platform data for reset gpio This patch adds platform data support for a reset GPIO. Also uses reset_gpio to toggle reset of the CODEC Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- include/sound/cs42l52.h | 2 ++ sound/soc/codecs/cs42l52.c | 21 +++++++++++++++++---- 2 files changed, 19 insertions(+), 4 deletions(-) diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index 4c68955f7330..7c2be4a51894 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -31,6 +31,8 @@ struct cs42l52_platform_data { /* Charge Pump Freq. Check datasheet Pg73 */ unsigned int chgfreq; + /* Reset GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b6fe4a..8367f3c571eb 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -1205,6 +1206,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l52_private *cs42l52; + struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1222,11 +1224,22 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } - i2c_set_clientdata(i2c_client, cs42l52); + if (pdata) + cs42l52->pdata = *pdata; - if (dev_get_platdata(&i2c_client->dev)) - memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), - sizeof(cs42l52->pdata)); + if (cs42l52->pdata.reset_gpio) { + ret = gpio_request_one(cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l52->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1); + } + + i2c_set_clientdata(i2c_client, cs42l52); ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, ARRAY_SIZE(cs42l52_threshold_patch)); -- cgit From 153723f6f1d13e7b9541b425ebdbaead4cc85346 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 25 Oct 2013 10:01:16 -0500 Subject: ASoC: cs42l52: convert pdata config to regmap_update_bits Moving platform data to bus probe and convert to regmap_update_bits. This will work nicer when converted to device tree instead of having it split into multiple probes Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 69 +++++++++++++++++++++++----------------------- 1 file changed, 34 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8367f3c571eb..56c5611fa752 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1117,40 +1117,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec) cs42l52->sysclk = CS42L52_DEFAULT_CLK; cs42l52->config.format = CS42L52_DEFAULT_FORMAT; - /* Set Platform MICx CFG */ - snd_soc_update_bits(codec, CS42L52_MICA_CTL, - CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.mica_cfg << - CS42L52_MIC_CTL_TYPE_SHIFT); - - snd_soc_update_bits(codec, CS42L52_MICB_CTL, - CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.micb_cfg << - CS42L52_MIC_CTL_TYPE_SHIFT); - - /* if Single Ended, Get Mic_Select */ - if (cs42l52->pdata.mica_cfg) - snd_soc_update_bits(codec, CS42L52_MICA_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.mica_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - if (cs42l52->pdata.micb_cfg) - snd_soc_update_bits(codec, CS42L52_MICB_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.micb_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - - /* Set Platform Charge Pump Freq */ - snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP, - CS42L52_CHARGE_PUMP_MASK, - cs42l52->pdata.chgfreq << - CS42L52_CHARGE_PUMP_SHIFT); - - /* Set Platform Bias Level */ - snd_soc_update_bits(codec, CS42L52_IFACE_CTL2, - CS42L52_IFACE_CTL2_BIAS_LVL, - cs42l52->pdata.micbias_lvl); - return ret; } @@ -1257,7 +1223,40 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } - regcache_cache_only(cs42l52->regmap, true); + /* Set Platform Data */ + if (cs42l52->pdata.mica_cfg) + regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.mica_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + if (cs42l52->pdata.micb_cfg) + regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.micb_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + if (cs42l52->pdata.mica_sel) + regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.mica_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + if (cs42l52->pdata.micb_sel) + regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.micb_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + + if (cs42l52->pdata.chgfreq) + regmap_update_bits(cs42l52->regmap, CS42L52_CHARGE_PUMP, + CS42L52_CHARGE_PUMP_MASK, + cs42l52->pdata.chgfreq << + CS42L52_CHARGE_PUMP_SHIFT); + + if (cs42l52->pdata.micbias_lvl) + regmap_update_bits(cs42l52->regmap, CS42L52_IFACE_CTL2, + CS42L52_IFACE_CTL2_BIAS_LVL, + cs42l52->pdata.micbias_lvl); ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l52, &cs42l52_dai, 1); -- cgit From e5f03af644c46b8713ddf5df545a32c1af8557ed Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 25 Oct 2013 10:01:17 -0500 Subject: ASoC: cs42l52: Add chip rev id message This patch adds a print message at bootup for the CODEC Rev ID Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 56c5611fa752..8b427c977083 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1223,6 +1223,9 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } + dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", + reg & 0xFF); + /* Set Platform Data */ if (cs42l52->pdata.mica_cfg) regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, -- cgit From f82bf8e2c8be73ebd719807bf054721930371174 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Oct 2013 18:06:09 +0200 Subject: ASoC: dmaengine: Use SNDRV_PCM_STREAM_LAST for array size ... to make the meaning more obvious. Signed-off-by: Takashi Iwai Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 0c469cbbe881..ee0790337ec9 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -25,7 +25,7 @@ #include struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; + struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; unsigned int flags; -- cgit From aab554ede931eddaca2e9b38c12489ae3f83fbe3 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 25 Oct 2013 10:01:15 -0500 Subject: ASoC: cs42l52: increase MAX_REGISTER for regmap_register_patch regmap_register_patch fails without the MAX_REGISTER set to highest register written to. Increase to register 0x47 Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012c4719..1a9412d86d17 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -269,6 +269,6 @@ #define CS42L52_FIX_BITS1 0x3E #define CS42L52_FIX_BITS2 0x47 -#define CS42L52_MAX_REGISTER 0x34 +#define CS42L52_MAX_REGISTER 0x47 #endif -- cgit From afb5a7793144179bfd856e3c928b69c43e2a1d8a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 26 Oct 2013 00:33:58 +0200 Subject: ALSA: hda - Sync EAPD with vmaster on AD1984A Thinkpads As EAPD on NID 0x12 (speaker pin) is used as the master amp on Thinkpads with AD1984A codec, we can hook this to vmaster for saving a bit more power at master mute state. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2aa2f579b4d6..87d2e0335ae4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -973,8 +973,11 @@ static void ad1884_fixup_thinkpad(struct hda_codec *codec, { struct ad198x_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.keep_eapd_on = 1; + spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + spec->eapd_nid = 0x12; + } } /* set magic COEFs for dmic */ -- cgit From a9c9cafdde46d06a28f92e3a68b5534fa268e92d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sat, 26 Oct 2013 15:31:26 +0100 Subject: ASoC: wm5110: Add missing routes for AEC Loopback Reported-by: Nariman Poushin Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd64384ca1c..8c91be5d67e3 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -983,24 +983,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "HPOUT3L", "OUT3L" }, + { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, { "HPOUT3R", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, + { "AEC Loopback", "SPKDAT2L", "OUT6L" }, + { "AEC Loopback", "SPKDAT2R", "OUT6R" }, { "SPKDAT2L", NULL, "OUT6L" }, { "SPKDAT2R", NULL, "OUT6R" }, -- cgit From 3d8f7318f929f3b84571ffac2ef7bf8bddfb6d41 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 25 Oct 2013 17:29:25 +0800 Subject: ASoC: fsl_spdif: fix return value check in fsl_spdif_probe() In case of error, the function platform_get_resource() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index c0fea02114e1..e1bf5ef31bdd 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1107,9 +1107,9 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (IS_ERR(res)) { + if (!res) { dev_err(&pdev->dev, "could not determine device resources\n"); - return PTR_ERR(res); + return -ENXIO; } regs = devm_ioremap_resource(&pdev->dev, res); -- cgit From a655f75c7502e5000c8745f458d3dbb9777f4aca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Oct 2013 16:33:12 +0200 Subject: ALSA: pcm_dmaengine: Remove hardcoded PCM formats Use the standard PCM helper function to figure out the sample bytes instead of hardcodec PCM format checks in snd_hwparams_to_dma_slave_config(). The patch also extends the format check for 8 bytes formats although no one should match so far. Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- sound/core/pcm_dmaengine.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index aa924d9b7986..94d08733cb38 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -63,23 +63,19 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { enum dma_slave_buswidth buswidth; + int bits; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + bits = snd_pcm_format_physical_width(params_format(params)); + if (bits < 8 || bits > 64) + return -EINVAL; + else if (bits == 8) buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; - break; - case SNDRV_PCM_FORMAT_S16_LE: + else if (bits == 16) buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; - break; - case SNDRV_PCM_FORMAT_S18_3LE: - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + else if (bits <= 32) buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - break; - default: - return -EINVAL; - } + else + buswidth = DMA_SLAVE_BUSWIDTH_8_BYTES; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->direction = DMA_MEM_TO_DEV; -- cgit From 63437313daaf90b372d5b383a8cc6ec8dce185fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 16:08:27 +0100 Subject: ALSA: memalloc: Yet another ifdef CONFIG_GENERIC_ALLOCATOR protection I obviously forgot to merge the right version... Reported-by: Stephen Rothwell Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 51a79218815b..809fd79ecad6 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -159,6 +159,7 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } +#ifdef CONFIG_GENERIC_ALLOCATOR /** * snd_malloc_dev_iram - allocate memory from on-chip internal ram * @dmab: buffer allocation record to store the allocated data @@ -198,6 +199,7 @@ void snd_free_dev_iram(struct snd_dma_buffer *dmab) if (pool && dmab->area) gen_pool_free(pool, (unsigned long)dmab->area, dmab->bytes); } +#endif /* CONFIG_GENERIC_ALLOCATOR */ #endif /* CONFIG_HAS_DMA */ /* -- cgit From 9b389a8a022110b4bc055a19b888283544d9eba6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 11:24:23 +0100 Subject: ALSA: 6fire: Fix probe of multiple cards The probe code of snd-usb-6fire driver overrides the devices[] pointer wrongly without checking whether it's already occupied or not. This would screw up the device disconnection later. Spotted by coverity CID 141423. Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/chip.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index c39c77978468..66edc4a7917f 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -101,7 +101,7 @@ static int usb6fire_chip_probe(struct usb_interface *intf, usb_set_intfdata(intf, chips[i]); mutex_unlock(®ister_mutex); return 0; - } else if (regidx < 0) + } else if (!devices[i] && regidx < 0) regidx = i; } if (regidx < 0) { -- cgit From 728deecdd4e9b4a58faf518638169c743c57cee2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 11:39:23 +0100 Subject: ALSA: hda - Fix possible NULL dereference in snd_hda_get_pin_label() Fix a possible NULL access of indexp in fill_audio_out_name() called from snd_hda_get_pin_label(). Spotted by coverity CID 402035. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 48a9d004d6d9..853c6a69e29e 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -638,7 +638,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, /* don't add channel suffix for Headphone controls */ int idx = get_hp_label_index(codec, nid, cfg->hp_pins, cfg->hp_outs); - if (idx >= 0) + if (idx >= 0 && indexp) *indexp = idx; sfx = ""; } -- cgit From 57d8ff617f871eee11bb2f648505592c2e859d0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 11:42:56 +0100 Subject: ALSA: hda - Add a fallthru comment ... to improve the readability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 20d065a17226..98bce9830be0 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -110,6 +110,7 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, case SND_BELL: if (hz) hz = 1000; + /* fallthru */ case SND_TONE: if (beep->linear_tone) beep->tone = beep_linear_tone(beep, hz); -- cgit From ca16ec02e1887cc88e9eb3504ee8ea6d8112d05d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 12:00:35 +0100 Subject: ALSA: hda - Remove locally dead codes Reported by coverity. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/patch_realtek.c | 2 -- 2 files changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0faab3b4dabb..de1a7670ba0d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2634,8 +2634,7 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, items = codec->mixers.list; for (i = 0; i < codec->mixers.used; i++) { struct snd_kcontrol *sctl = items[i].kctl; - if (!sctl || !sctl->id.name || - sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER) + if (!sctl || sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER) continue; for (s = slaves; *s; s++) { char tmpname[sizeof(sctl->id.name)]; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1f0a040a9818..894703532ab7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -554,8 +554,6 @@ do_sku: nid = portd; else if (tmp == 3) nid = porti; - else - return 1; if (found_in_nid_list(nid, spec->gen.autocfg.line_out_pins, spec->gen.autocfg.line_outs)) return 1; -- cgit From 1f96153bc613c526f5ea39a7f366b3bdf70249be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 12:40:46 +0100 Subject: ALSA: pcm: Add fallthru comments Just to improve readability. Spotted by coverity CID 115002 and 115003. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b71be579c6ec..01a5e05ede95 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2428,6 +2428,7 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream) case SNDRV_PCM_STATE_DRAINING: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) goto __badfd; + /* Fall through */ case SNDRV_PCM_STATE_RUNNING: if ((err = snd_pcm_update_hw_ptr(substream)) < 0) break; @@ -2460,6 +2461,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DRAINING: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) goto __badfd; + /* Fall through */ case SNDRV_PCM_STATE_RUNNING: if ((err = snd_pcm_update_hw_ptr(substream)) < 0) break; -- cgit From 60f6fef877d52525b8887d27cfacf74a279e3e12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 12:54:52 +0100 Subject: ALSA: Optimize module name check module->name is a fixed array, so we can check the empty contents straightforwardly in module_slot_match(). Spotted by coverity CID 1056786. Signed-off-by: Takashi Iwai --- sound/core/init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/init.c b/sound/core/init.c index 6b9087115da2..01a89383a062 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -66,7 +66,7 @@ static int module_slot_match(struct module *module, int idx) #ifdef MODULE const char *s1, *s2; - if (!module || !module->name || !slots[idx]) + if (!module || !*module->name || !slots[idx]) return 0; s1 = module->name; -- cgit From b327d25c1c3d475d1d1217be520801283e8bdf29 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Tue, 29 Oct 2013 12:05:02 +0900 Subject: ALSA: Fix typo in documentation/alsa Correct spelling typo in documentation/alsa Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- Documentation/sound/alsa/Audiophile-Usb.txt | 2 +- Documentation/sound/alsa/CMIPCI.txt | 2 +- Documentation/sound/alsa/compress_offload.txt | 6 +++--- Documentation/sound/alsa/soc/DPCM.txt | 4 ++-- Documentation/sound/alsa/soc/dapm.txt | 4 ++-- 6 files changed, 10 insertions(+), 10 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 95731a08f257..b8dd0df76952 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -616,7 +616,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. As default, snd-dummy drivers doesn't allocate the real buffers but either ignores read/write or mmap a single dummy page to all - buffer pages, in order to save the resouces. If your apps need + buffer pages, in order to save the resources. If your apps need the read/ written buffer data to be consistent, pass fake_buffer=0 option. diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index 654dd3b694a8..e7a5ed4dcae8 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -232,7 +232,7 @@ The parameter can be given: # modprobe snd-usb-audio index=1 device_setup=0x09 * Or while configuring the modules options in your modules configuration file - (tipically a .conf file in /etc/modprobe.d/ directory: + (typically a .conf file in /etc/modprobe.d/ directory: alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 device_setup=0x09 diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt index 16935c8561f7..4e36e6e809ca 100644 --- a/Documentation/sound/alsa/CMIPCI.txt +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -87,7 +87,7 @@ with 4 channels, and use the interleaved 4 channel data. -There are some control switchs affecting to the speaker connections: +There are some control switches affecting to the speaker connections: "Line-In Mode" - an enum control to change the behavior of line-in jack. Either "Line-In", "Rear Output" or "Bass Output" can diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index fd74ff26376e..630c492c3dc2 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -217,12 +217,12 @@ Not supported: would be enabled with ALSA kcontrols. - Audio policy/resource management. This API does not provide any - hooks to query the utilization of the audio DSP, nor any premption + hooks to query the utilization of the audio DSP, nor any preemption mechanisms. -- No notion of underun/overrun. Since the bytes written are compressed +- No notion of underrun/overrun. Since the bytes written are compressed in nature and data written/read doesn't translate directly to - rendered output in time, this does not deal with underrun/overun and + rendered output in time, this does not deal with underrun/overrun and maybe dealt in user-library Credits: diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt index aa8546f2d144..0110180b7ac6 100644 --- a/Documentation/sound/alsa/soc/DPCM.txt +++ b/Documentation/sound/alsa/soc/DPCM.txt @@ -192,7 +192,7 @@ This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets the "no_pcm" flag to mark it has a BE and sets flags for supported stream directions using "dpcm_playback" and "dpcm_capture" above. -The BE has also flags set for ignoreing suspend and PM down time. This allows +The BE has also flags set for ignoring suspend and PM down time. This allows the BE to work in a hostless mode where the host CPU is not transferring data like a BT phone call :- @@ -328,7 +328,7 @@ The host can control the hostless link either by :- between both DAIs. 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM - graph. Control is then carried out by the FE as regualar PCM operations. + graph. Control is then carried out by the FE as regular PCM operations. This method gives more control over the DAI links, but requires much more userspace code to control the link. Its recommended to use CODEC<->CODEC unless your HW needs more fine grained sequencing of the PCM ops. diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 7dfd88ce31ac..6faab4880006 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -30,7 +30,7 @@ There are 4 power domains within DAPM machine driver and responds to asynchronous events e.g when HP are inserted - 3. Path domain - audio susbsystem signal paths + 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. @@ -64,7 +64,7 @@ Audio DAPM widgets fall into a number of types:- o Speaker - Speaker o Supply - Power or clock supply widget used by other widgets. o Regulator - External regulator that supplies power to audio components. - o Clock - External clock that supplies clock to audio componnents. + o Clock - External clock that supplies clock to audio components. o AIF IN - Audio Interface Input (with TDM slot mask). o AIF OUT - Audio Interface Output (with TDM slot mask). o Siggen - Signal Generator. -- cgit From 9f694bc7936a7e4e9c9efac2900cddaf71303c0a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 11:56:21 +0100 Subject: ALSA: memalloc: Make snd_{malloc|free}_dev_iram() static These are used only locally. Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 809fd79ecad6..278248b8f22a 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -167,7 +167,7 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, * * This function requires iram phandle provided via of_node */ -void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) +static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) { struct device *dev = dmab->dev.dev; struct gen_pool *pool = NULL; @@ -192,7 +192,7 @@ void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) * snd_free_dev_iram - free allocated specific memory from on-chip internal ram * @dmab: buffer allocation record to store the allocated data */ -void snd_free_dev_iram(struct snd_dma_buffer *dmab) +static void snd_free_dev_iram(struct snd_dma_buffer *dmab) { struct gen_pool *pool = dmab->private_data; -- cgit From a40a3937222c728be925f2d78650cfe9b20be3f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 11:59:31 +0100 Subject: ALSA: memalloc: NULL-initialize in snd_malloc_dev_iram() dmab->area and addr fields should be cleared at the head of snd_malloc_dev_iram() as especially dmab->area is used to indicate the allocation failure / fallback. Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 278248b8f22a..9d93f02c6285 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -172,6 +172,9 @@ static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) struct device *dev = dmab->dev.dev; struct gen_pool *pool = NULL; + dmab->area = NULL; + dmab->addr = 0; + if (dev->of_node) pool = of_get_named_gen_pool(dev->of_node, "iram", 0); -- cgit From a8d09e3a6dfc44fd2246f3c57d2cd00e1a56e2bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 15:07:19 +0100 Subject: ALSA: opl3: Fix possible negative array index access Spotted by coverity CID 115196. Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_midi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 0c796bcbc0a3..6c6d09a51f42 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -390,6 +390,11 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) voice = snd_opl3_oss_map[chan->number]; } + if (voice < 0) { + spin_unlock_irqrestore(&opl3->voice_lock, flags); + return; + } + if (voice < MAX_OPL2_VOICES) { /* Left register block for voices 0 .. 8 */ reg_side = OPL3_LEFT; -- cgit From 75415df8ffbac2ace9463ceffb3f21299af2548a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 15:15:20 +0100 Subject: ALSA: pcsp: Fix initialization with nopcm=1 When nopcm=1 is set, some initializations based on hrtimer resolution might be bogus because the driver checks the resolution only when nopcm=0. Simply get the resolution always at first for fixing the bug. Spotted by coverity CID 139740. Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 1c19cd7ad26e..f664bae3b9b0 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -46,8 +46,9 @@ static int snd_pcsp_create(struct snd_card *card) int err; int div, min_div, order; + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (!nopcm) { - hrtimer_get_res(CLOCK_MONOTONIC, &tp); if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { printk(KERN_ERR "PCSP: Timer resolution is not sufficient " "(%linS)\n", tp.tv_nsec); -- cgit From 97f44f56ca94709f45bc348f5d2c88696eae5f9b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 15:20:06 +0100 Subject: ALSA: Limit the fallback card id string size When no proper id string is given, the driver tries to fall back to copy the proc_root name string via strcpy(), but this might overflow the fixed string size. Let's use strlcpy(). Spotted by coverity CID 139008. Signed-off-by: Takashi Iwai --- sound/core/init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/init.c b/sound/core/init.c index 01a89383a062..1351f22f651c 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -597,7 +597,7 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, /* last resort... */ snd_printk(KERN_ERR "unable to set card id (%s)\n", id); if (card->proc_root->name) - strcpy(card->id, card->proc_root->name); + strlcpy(card->id, card->proc_root->name, sizeof(card->id)); } /** -- cgit From 57a4451d26eef4ccbf3b32fd116295f001c18cb4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 15:26:12 +0100 Subject: ALSA: Use strlcpy() instead of strncpy() We tend to make stupid mistakes with strncpy(). Let's take a safer one, strlcpy(). Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 2 +- sound/i2c/other/ak4xxx-adda.c | 2 +- sound/pci/cs5535audio/cs5535audio_olpc.c | 4 ++-- sound/pci/ice1712/psc724.c | 4 ++-- sound/pci/ice1712/wm8776.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- 6 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5066a3768b28..9a2ac1e0f77a 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -185,7 +185,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev) goto err; card->dev = &dev->dev; - strncpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); + strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm); if (ret) diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index ed726d1569e8..f3735e64791c 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -583,7 +583,7 @@ static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, if (idx >= num_names) return -EINVAL; input_names = ak->adc_info[mixer_ch].input_names; - strncpy(uinfo->value.enumerated.name, input_names[idx], + strlcpy(uinfo->value.enumerated.name, input_names[idx], sizeof(uinfo->value.enumerated.name)); return 0; } diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c index da1cb9c4c76c..e6a44507d557 100644 --- a/sound/pci/cs5535audio/cs5535audio_olpc.c +++ b/sound/pci/cs5535audio/cs5535audio_olpc.c @@ -161,13 +161,13 @@ int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97) /* drop the original AD1888 HPF control */ memset(&elem, 0, sizeof(elem)); elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strncpy(elem.name, "High Pass Filter Enable", sizeof(elem.name)); + strlcpy(elem.name, "High Pass Filter Enable", sizeof(elem.name)); snd_ctl_remove_id(card, &elem); /* drop the original V_REFOUT control */ memset(&elem, 0, sizeof(elem)); elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strncpy(elem.name, "V_REFOUT Enable", sizeof(elem.name)); + strlcpy(elem.name, "V_REFOUT Enable", sizeof(elem.name)); snd_ctl_remove_id(card, &elem); /* add the OLPC-specific controls */ diff --git a/sound/pci/ice1712/psc724.c b/sound/pci/ice1712/psc724.c index 302ac6ddd545..4019cf27d117 100644 --- a/sound/pci/ice1712/psc724.c +++ b/sound/pci/ice1712/psc724.c @@ -203,12 +203,12 @@ static void psc724_set_jack_state(struct snd_ice1712 *ice, bool hp_connected) /* notify about master speaker mute change */ memset(&elem_id, 0, sizeof(elem_id)); elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strncpy(elem_id.name, "Master Speakers Playback Switch", + strlcpy(elem_id.name, "Master Speakers Playback Switch", sizeof(elem_id.name)); kctl = snd_ctl_find_id(ice->card, &elem_id); snd_ctl_notify(ice->card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); /* and headphone mute change */ - strncpy(elem_id.name, spec->wm8776.ctl[WM8776_CTL_HP_SW].name, + strlcpy(elem_id.name, spec->wm8776.ctl[WM8776_CTL_HP_SW].name, sizeof(elem_id.name)); kctl = snd_ctl_find_id(ice->card, &elem_id); snd_ctl_notify(ice->card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index a3c05fe5daf9..5227cb08247f 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -52,7 +52,7 @@ static void snd_wm8776_activate_ctl(struct snd_wm8776 *wm, unsigned int index_offset; memset(&elem_id, 0, sizeof(elem_id)); - strncpy(elem_id.name, ctl_name, sizeof(elem_id.name)); + strlcpy(elem_id.name, ctl_name, sizeof(elem_id.name)); elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; kctl = snd_ctl_find_id(card, &elem_id); if (!kctl) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 2907e68150cb..e98dc008de0b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6400,7 +6400,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, memset(&hdspm_version, 0, sizeof(hdspm_version)); hdspm_version.card_type = hdspm->io_type; - strncpy(hdspm_version.cardname, hdspm->card_name, + strlcpy(hdspm_version.cardname, hdspm->card_name, sizeof(hdspm_version.cardname)); hdspm_version.serial = hdspm->serial; hdspm_version.firmware_rev = hdspm->firmware_rev; -- cgit From 9bd0f5c042e218c0573e730d181287abecd6b18b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 15:39:37 +0100 Subject: ALSA: ad1889: Fix right attenuation proc output The right attenuation bits aren't needed to be shifted. Spotted by coverity CID 11427. Signed-off-by: Takashi Iwai --- sound/pci/ad1889.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index d2b9d617aee5..b680d03e2419 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -739,7 +739,7 @@ snd_ad1889_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffe reg = ad1889_readw(chip, AD_DS_WADA); snd_iprintf(buffer, "Right: %s, -%d dB\n", (reg & AD_DS_WADA_RWAM) ? "mute" : "unmute", - ((reg & AD_DS_WADA_RWAA) >> 8) * 3); + (reg & AD_DS_WADA_RWAA) * 3); reg = ad1889_readw(chip, AD_DS_WAS); snd_iprintf(buffer, "Wave samplerate: %u Hz\n", reg); -- cgit From 329bbd9ba738b85675151b82260cd08bdd765de2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 15:42:29 +0100 Subject: ALSA: ali5451: Drop unused variable The variable runtime is never used, and this might be even a source of NULL-dereference. Nothing better than killing it. Spotted by coverity CID 100862. Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 3dfa12b670eb..c6835a3d64fb 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -855,7 +855,6 @@ static void snd_ali_disable_spdif_out(struct snd_ali *codec) static void snd_ali_update_ptr(struct snd_ali *codec, int channel) { struct snd_ali_voice *pvoice; - struct snd_pcm_runtime *runtime; struct snd_ali_channel_control *pchregs; unsigned int old, mask; #ifdef ALI_DEBUG @@ -872,7 +871,6 @@ static void snd_ali_update_ptr(struct snd_ali *codec, int channel) return; pvoice = &codec->synth.voices[channel]; - runtime = pvoice->substream->runtime; udelay(100); spin_lock(&codec->reg_lock); -- cgit From 2026d24ef2ea8caad5e87662a58075e930ccab63 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 16:04:37 +0100 Subject: ALSA: rme96: Return error code in PCM copy ops Just pass the error code returned from copy_from_user_toio() and copy_to_user_fromio() helpers. Spotted by coverity CID 114119. Signed-off-by: Takashi Iwai --- sound/pci/rme96.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index bb9ebc5543d7..0236363c301f 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -350,9 +350,8 @@ snd_rme96_playback_copy(struct snd_pcm_substream *substream, struct rme96 *rme96 = snd_pcm_substream_chip(substream); count <<= rme96->playback_frlog; pos <<= rme96->playback_frlog; - copy_from_user_toio(rme96->iobase + RME96_IO_PLAY_BUFFER + pos, src, - count); - return 0; + return copy_from_user_toio(rme96->iobase + RME96_IO_PLAY_BUFFER + pos, src, + count); } static int @@ -365,9 +364,8 @@ snd_rme96_capture_copy(struct snd_pcm_substream *substream, struct rme96 *rme96 = snd_pcm_substream_chip(substream); count <<= rme96->capture_frlog; pos <<= rme96->capture_frlog; - copy_to_user_fromio(dst, rme96->iobase + RME96_IO_REC_BUFFER + pos, - count); - return 0; + return copy_to_user_fromio(dst, rme96->iobase + RME96_IO_REC_BUFFER + pos, + count); } /* -- cgit From e12483e0f3dbc32dad8fa1dc97efac22b6aee94f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 16:37:11 +0100 Subject: ALSA: ak4114: Fix wrong register array size The size of the register cache array is actually 6 instead of 7, as it caches up to AK4114_REG_INT1_MASK. This resulted in unexpected access out of array range, although most of them aren't so serious (just reading one more byte on the stack at snd_ak4114_create()). Also, the check of cache size was wrongly done by checking with sizeof() instead of ARRAY_SIZE(). Fixed this together. (And yes, hardcoded numbers are bad, but I keep the coding style as is for making it clear what this patch actually does.) Spotted by coverity among several CIDs, e.g. 711621. Signed-off-by: Takashi Iwai --- include/sound/ak4114.h | 4 ++-- sound/i2c/other/ak4114.c | 8 ++++---- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h index 3ce69fd92523..52f02a60dba7 100644 --- a/include/sound/ak4114.h +++ b/include/sound/ak4114.h @@ -170,7 +170,7 @@ struct ak4114 { void * private_data; unsigned int init: 1; spinlock_t lock; - unsigned char regmap[7]; + unsigned char regmap[6]; unsigned char txcsb[5]; struct snd_kcontrol *kctls[AK4114_CONTROLS]; struct snd_pcm_substream *playback_substream; @@ -189,7 +189,7 @@ struct ak4114 { int snd_ak4114_create(struct snd_card *card, ak4114_read_t *read, ak4114_write_t *write, - const unsigned char pgm[7], const unsigned char txcsb[5], + const unsigned char pgm[6], const unsigned char txcsb[5], void *private_data, struct ak4114 **r_ak4114); void snd_ak4114_reg_write(struct ak4114 *ak4114, unsigned char reg, unsigned char mask, unsigned char val); void snd_ak4114_reinit(struct ak4114 *ak4114); diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 5bf4fca19e48..15ae0250eace 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -60,7 +60,7 @@ static void reg_dump(struct ak4114 *ak4114) printk(KERN_DEBUG "AK4114 REG DUMP:\n"); for (i = 0; i < 0x20; i++) - printk(KERN_DEBUG "reg[%02x] = %02x (%02x)\n", i, reg_read(ak4114, i), i < sizeof(ak4114->regmap) ? ak4114->regmap[i] : 0); + printk(KERN_DEBUG "reg[%02x] = %02x (%02x)\n", i, reg_read(ak4114, i), i < ARRAY_SIZE(ak4114->regmap) ? ak4114->regmap[i] : 0); } #endif @@ -81,7 +81,7 @@ static int snd_ak4114_dev_free(struct snd_device *device) int snd_ak4114_create(struct snd_card *card, ak4114_read_t *read, ak4114_write_t *write, - const unsigned char pgm[7], const unsigned char txcsb[5], + const unsigned char pgm[6], const unsigned char txcsb[5], void *private_data, struct ak4114 **r_ak4114) { struct ak4114 *chip; @@ -101,7 +101,7 @@ int snd_ak4114_create(struct snd_card *card, chip->private_data = private_data; INIT_DELAYED_WORK(&chip->work, ak4114_stats); - for (reg = 0; reg < 7; reg++) + for (reg = 0; reg < 6; reg++) chip->regmap[reg] = pgm[reg]; for (reg = 0; reg < 5; reg++) chip->txcsb[reg] = txcsb[reg]; @@ -142,7 +142,7 @@ static void ak4114_init_regs(struct ak4114 *chip) /* release reset, but leave powerdown */ reg_write(chip, AK4114_REG_PWRDN, (old | AK4114_RST) & ~AK4114_PWN); udelay(200); - for (reg = 1; reg < 7; reg++) + for (reg = 1; reg < 6; reg++) reg_write(chip, reg, chip->regmap[reg]); for (reg = 0; reg < 5; reg++) reg_write(chip, reg + AK4114_REG_TXCSB0, chip->txcsb[reg]); -- cgit From 4c88b7f287e329698727049641ecdf50c8c66af5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 16:45:34 +0100 Subject: ALSA: ice1724: Fix uninitialized variable access Spotted by coverity CIDs 751505 and 751506. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wm8766.c | 3 ++- sound/pci/ice1712/wm8776.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index e473f8a88f9c..21b373b2e260 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -253,7 +253,8 @@ static int snd_wm8766_ctl_get(struct snd_kcontrol *kcontrol, } if (wm->ctl[n].flags & WM8766_FLAG_INVERT) { val1 = wm->ctl[n].max - (val1 - wm->ctl[n].min); - val2 = wm->ctl[n].max - (val2 - wm->ctl[n].min); + if (wm->ctl[n].flags & WM8766_FLAG_STEREO) + val2 = wm->ctl[n].max - (val2 - wm->ctl[n].min); } ucontrol->value.integer.value[0] = val1; if (wm->ctl[n].flags & WM8766_FLAG_STEREO) diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index 5227cb08247f..e66c0da62014 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -526,7 +526,8 @@ static int snd_wm8776_ctl_get(struct snd_kcontrol *kcontrol, } if (wm->ctl[n].flags & WM8776_FLAG_INVERT) { val1 = wm->ctl[n].max - (val1 - wm->ctl[n].min); - val2 = wm->ctl[n].max - (val2 - wm->ctl[n].min); + if (wm->ctl[n].flags & WM8776_FLAG_STEREO) + val2 = wm->ctl[n].max - (val2 - wm->ctl[n].min); } ucontrol->value.integer.value[0] = val1; if (wm->ctl[n].flags & WM8776_FLAG_STEREO) -- cgit From 9cd5ab9c3c32767e5c5e76bed9b0e2b84512f936 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 16:51:36 +0100 Subject: ALSA: lola: Fix uninitialized variable access in error message The FUNCTION_TYPE parameter isn't associated with any NID, thus showing the uninitialized nid in the error message is simply nonsense. Spotted by coverity CID 145068. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 7307d97186cb..0568540dc8d3 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -463,7 +463,7 @@ static int lola_parse_tree(struct lola *chip) err = lola_read_param(chip, 1, LOLA_PAR_FUNCTION_TYPE, &val); if (err < 0) { - printk(KERN_ERR SFX "Can't read FUNCTION_TYPE for 0x%x\n", nid); + printk(KERN_ERR SFX "Can't read FUNCTION_TYPE\n"); return err; } if (val != 1) { -- cgit From b5ef3f2a8074af7aef9a32f4535c57f986364a60 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 29 Oct 2013 17:06:27 +0800 Subject: ASoC: wm8962: Fix null pointer pdata access in I2C probe() When using DT binding to pass private data, there would be Kernel panic occuring due to NULL pointer access in wm8962_i2c_probe(). Thus fix it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2bf9ee7c5407..7dd79c4efc13 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3616,28 +3616,28 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, 0); /* Apply static configuration for GPIOs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) - if (pdata->gpio_init[i]) { + for (i = 0; i < ARRAY_SIZE(wm8962->pdata.gpio_init); i++) + if (wm8962->pdata.gpio_init[i]) { wm8962_set_gpio_mode(wm8962, i + 1); regmap_write(wm8962->regmap, 0x200 + i, - pdata->gpio_init[i] & 0xffff); + wm8962->pdata.gpio_init[i] & 0xffff); } /* Put the speakers into mono mode? */ - if (pdata->spk_mono) + if (wm8962->pdata.spk_mono) regmap_update_bits(wm8962->regmap, WM8962_CLASS_D_CONTROL_2, WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); /* Micbias setup, detection enable and detection * threasholds. */ - if (pdata->mic_cfg) + if (wm8962->pdata.mic_cfg) regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, WM8962_MICDET_ENA | WM8962_MICDET_THR_MASK | WM8962_MICSHORT_THR_MASK | WM8962_MICBIAS_LVL, - pdata->mic_cfg); + wm8962->pdata.mic_cfg); /* Latch volume update bits */ regmap_update_bits(wm8962->regmap, WM8962_LEFT_INPUT_VOLUME, @@ -3682,7 +3682,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, } if (wm8962->irq) { - if (pdata->irq_active_low) { + if (wm8962->pdata.irq_active_low) { trigger = IRQF_TRIGGER_LOW; irq_pol = WM8962_IRQ_POL; } else { -- cgit From d2e92709e88d97c001b6bb96054ecb06d99d0dc6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 07:50:53 +0100 Subject: ALSA: hda - Disable AA-loopback on ALC283 Chromebook ALC283-based Chromebook suffers from occasional white noise, and it turned out that this comes from AA-loopback. Disable this output path by just clearing mixer_nid, then the generic parser will skip the creation of AA-loopback path. Reported-and-tested-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 894703532ab7..c516fa93ee63 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3445,6 +3445,8 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: alc283_chromebook_caps(codec); + /* Disable AA-loopback as it causes white noise */ + spec->gen.mixer_nid = 0; spec->gen.hp_automute_hook = alc283_hp_automute_hook; /* MIC2-VREF control */ /* Set to manual mode */ -- cgit From 00ecdd93a80fda1336bf5413b1d705c742a5b598 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 08:34:59 +0100 Subject: ASoC: ab8500: Add missing of NULL check of devm_kzalloc() Spotted by coverity CID 712316. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 7f6ca111659b..10be4cbfe969 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2570,6 +2570,8 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) /* Create driver private-data struct */ drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; drvdata->sid_status = SID_UNCONFIGURED; drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); -- cgit From 166a34d27fcad1eeb0322cff23939a1910f8a77c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 08:35:01 +0100 Subject: ASoC: ab8500: Fix invalid cast to long pointer Don't cast to long pointers blindly just for using find_first_bit() and co. This is certainly not portable at all. Reimplement the code with ffs() and fls() instead. This is a slight optimization, too. Spotted by coverity CID 1056484 and 1056485. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 10be4cbfe969..3ef481551740 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2295,17 +2295,17 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, case 0: break; case 1: - slot = find_first_bit((unsigned long *)&tx_mask, 32); + slot = ffs(tx_mask); snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; case 2: - slot = find_first_bit((unsigned long *)&tx_mask, 32); + slot = ffs(tx_mask); snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); - slot = find_next_bit((unsigned long *)&tx_mask, 32, slot + 1); + slot = fls(tx_mask); snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; @@ -2336,18 +2336,18 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, case 0: break; case 1: - slot = find_first_bit((unsigned long *)&rx_mask, 32); + slot = ffs(rx_mask); snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), AB8500_MASK_SLOT(slot), AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); break; case 2: - slot = find_first_bit((unsigned long *)&rx_mask, 32); + slot = ffs(rx_mask); snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), AB8500_MASK_SLOT(slot), AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); - slot = find_next_bit((unsigned long *)&rx_mask, 32, slot + 1); + slot = fls(rx_mask); snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), AB8500_MASK_SLOT(slot), -- cgit From c36c89096cb9f95fbdb0a6f3d80d4b9a50537ed3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 08:35:03 +0100 Subject: ASoC: wm0010: Fix possible out-of-bounds array read Spotted by coverity CID 744701. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index d5ebcb00019b..bf7804a12863 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -793,11 +793,11 @@ static int wm0010_set_sysclk(struct snd_soc_codec *codec, int source, wm0010->max_spi_freq = 0; } else { for (i = 0; i < ARRAY_SIZE(pll_clock_map); i++) - if (freq >= pll_clock_map[i].max_sysclk) + if (freq >= pll_clock_map[i].max_sysclk) { + wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed; + wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1; break; - - wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed; - wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1; + } } return 0; -- cgit From 8e38395360844806041ea69ab9690f5f174bc40c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 17:41:12 +0100 Subject: ALSA: hda - Add extra chmap for 2.1 outputs on ASUS laptops ASUS N56VZ and N76VZ laptops have a bass speaker but its output comes only from the right channel. This patch adds the extra chmap specific to these models. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=846531 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 13ad49ca9357..2d7c86d81196 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4426,6 +4426,25 @@ static void alc272_fixup_mario(struct hda_codec *codec, "hda_codec: failed to override amp caps for NID 0x2\n"); } +static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_NA, SNDRV_CHMAP_LFE } }, /* LFE only on right */ + { } +}; + +/* override the 2.1 chmap */ +static void alc662_fixup_bass_chmap(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_BUILD) { + struct alc_spec *spec = codec->spec; + spec->gen.pcm_rec[0].stream[0].chmap = asus_pcm_2_1_chmaps; + } +} + enum { ALC662_FIXUP_ASPIRE, ALC662_FIXUP_IDEAPAD, @@ -4446,6 +4465,7 @@ enum { ALC662_FIXUP_INV_DMIC, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_HEADSET_MODE, + ALC662_FIXUP_BASS_CHMAP, }; static const struct hda_fixup alc662_fixups[] = { @@ -4620,6 +4640,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc668, }, + [ALC662_FIXUP_BASS_CHMAP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc662_fixup_bass_chmap, + .chained = true, + .chain_id = ALC662_FIXUP_ASUS_MODE4 + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -4633,8 +4659,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), - SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_ASUS_MODE4), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- cgit From fe329a1a92cfe2d0c7e04fe3bc63761dc0f35950 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 08:35:07 +0100 Subject: ASoC: wm8996: Fix negative array index read Spotted by coverity CID 146355. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 46fe83d2b224..b70379ebd142 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -438,6 +438,8 @@ static int wm8996_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int block = wm8996_get_retune_mobile_block(kcontrol->id.name); + if (block < 0) + return block; ucontrol->value.enumerated.item[0] = wm8996->retune_mobile_cfg[block]; return 0; -- cgit From 5a7615cf1fcaaf1598b5689e54915d88c2344788 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 08:35:06 +0100 Subject: ASoC: rt5640: Fix ignored error checks The negative error value returned from get_sdp_info() is ignored because it's assigned to unsigned variables. Spotted by coverity CIDs 1042657, 1042658. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4d041d376f31..a3fb41179636 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1604,8 +1604,8 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk, dai_sel; - int pre_div, bclk_ms, frame_size; + unsigned int val_len = 0, val_clk, mask_clk; + int dai_sel, pre_div, bclk_ms, frame_size; rt5640->lrck[dai->id] = params_rate(params); pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); @@ -1675,7 +1675,8 @@ static int rt5640_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0, dai_sel; + unsigned int reg_val = 0; + int dai_sel; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: -- cgit From 8b4b30365ce6cefe4193f439ac7263bb2cdd66fd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2013 18:40:04 +0100 Subject: ASoC: ml26124: Fix negative array index read get_coeff() may return an error. Spotted by coverity CID 703394. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/ml26124.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 26118828782b..185fa3bc3052 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -342,6 +342,8 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream, struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); int i = get_coeff(priv->mclk, params_rate(hw_params)); + if (i < 0) + return i; priv->substream = substream; priv->rate = params_rate(hw_params); -- cgit From 9ade09d6c62e48fba6c74ce3958ca1035dfd8427 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 29 Oct 2013 00:52:19 -0700 Subject: ASoC: rcar: remove original filter from rsnd_dma_init() Remove original filter from rsnd_dma_init(), and use SH-DMA suitable filter. This new style can be used from Device Tree. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 31 ++++++++++++++++++++----------- 1 file changed, 20 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index b234ed663073..78c35b44fc04 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -94,6 +94,7 @@ * */ #include +#include #include "rsnd.h" #define RSND_RATES SNDRV_PCM_RATE_8000_96000 @@ -209,13 +210,6 @@ int rsnd_dma_available(struct rsnd_dma *dma) return !!dma->chan; } -static bool rsnd_dma_filter(struct dma_chan *chan, void *param) -{ - chan->private = param; - - return true; -} - int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int is_play, int id, int (*inquiry)(struct rsnd_dma *dma, @@ -223,7 +217,9 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int (*complete)(struct rsnd_dma *dma)) { struct device *dev = rsnd_priv_to_dev(priv); + struct dma_slave_config cfg; dma_cap_mask_t mask; + int ret; if (dma->chan) { dev_err(dev, "it already has dma channel\n"); @@ -233,15 +229,23 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - dma->slave.shdma_slave.slave_id = id; - - dma->chan = dma_request_channel(mask, rsnd_dma_filter, - &dma->slave.shdma_slave); + dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, + (void *)id, dev, + is_play ? "tx" : "rx"); if (!dma->chan) { dev_err(dev, "can't get dma channel\n"); return -EIO; } + cfg.slave_id = id; + cfg.dst_addr = 0; /* use default addr when playback */ + cfg.src_addr = 0; /* use default addr when capture */ + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + ret = dmaengine_slave_config(dma->chan, &cfg); + if (ret < 0) + goto rsnd_dma_init_err; + dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; dma->priv = priv; dma->inquiry = inquiry; @@ -249,6 +253,11 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, INIT_WORK(&dma->work, rsnd_dma_do_work); return 0; + +rsnd_dma_init_err: + rsnd_dma_quit(priv, dma); + + return ret; } void rsnd_dma_quit(struct rsnd_priv *priv, -- cgit From a19685cb72bb6a80ac453e76b3ab3bb7770e1742 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2013 14:21:46 +0100 Subject: ASoC: Use strlcpy() for copying in snd_soc_info_enum_double() The provided texts aren't guaranteed to be in the fixed size. Spotted by coverity CID 139318. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0860a7f11299..b38e0ee622df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2552,8 +2552,9 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, if (uinfo->value.enumerated.item > e->max - 1) uinfo->value.enumerated.item = e->max - 1; - strcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item]); + strlcpy(uinfo->value.enumerated.name, + e->texts[uinfo->value.enumerated.item], + sizeof(uinfo->value.enumerated.name)); return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); -- cgit From c5914b0aaea6494aaa9e415cbd32f8b7eb604af0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Oct 2013 17:47:39 -0700 Subject: ASoC: pcm: Check for ops before deferencing them Ensure that we always check that an ops structure is present before we try to use it, improving the robustness of the system. Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 43 ++++++++++++++++++++++--------------------- 1 file changed, 22 insertions(+), 21 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d4498723b375..d81b79251760 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -190,7 +190,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->startup) { ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: can't open interface" @@ -208,7 +208,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops->startup) { + if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { dev_err(codec_dai->dev, "ASoC: can't open codec" @@ -463,7 +463,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops->prepare) { + if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n", @@ -472,7 +472,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (cpu_dai->driver->ops->prepare) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", @@ -523,7 +523,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->driver->ops->hw_params) { + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) { ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); if (ret < 0) { dev_err(codec_dai->dev, "ASoC: can't set %s hw params:" @@ -532,7 +532,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (cpu_dai->driver->ops->hw_params) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) { ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: %s hw params failed: %d\n", @@ -559,11 +559,11 @@ out: return ret; platform_err: - if (cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_err: @@ -600,10 +600,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->driver->ops->hw_free(substream); /* now free hw params for the DAIs */ - if (codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); - if (cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); mutex_unlock(&rtd->pcm_mutex); @@ -618,7 +618,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->driver->ops->trigger) { + if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; @@ -630,7 +630,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->driver->ops->trigger) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->trigger) { ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -647,19 +647,20 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->driver->ops->bespoke_trigger) { + if (codec_dai->driver->ops && + codec_dai->driver->ops->bespoke_trigger) { ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } - if (platform->driver->bespoke_trigger) { + if (platform->driver->ops && platform->driver->bespoke_trigger) { ret = platform->driver->bespoke_trigger(substream, cmd); if (ret < 0) return ret; } - if (cpu_dai->driver->ops->bespoke_trigger) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->bespoke_trigger) { ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -684,10 +685,10 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->pointer) offset = platform->driver->ops->pointer(substream); - if (cpu_dai->driver->ops->delay) + if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->driver->ops->delay) + if (codec_dai->driver->ops && codec_dai->driver->ops->delay) delay += codec_dai->driver->ops->delay(substream, codec_dai); if (platform->driver->delay) @@ -1673,7 +1674,7 @@ static int soc_pcm_ioctl(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; - if (platform->driver->ops->ioctl) + if (platform->driver->ops && platform->driver->ops->ioctl) return platform->driver->ops->ioctl(substream, cmd, arg); return snd_pcm_lib_ioctl(substream, cmd, arg); } @@ -1934,8 +1935,8 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) dev_dbg(be->dev, "ASoC: BE digital mute %s\n", be->dai_link->name); - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, mute); + if (drv->ops && drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, mute); } return 0; @@ -2224,7 +2225,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); int snd_soc_platform_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_platform *platform) { - if (platform->driver->ops->trigger) + if (platform->driver->ops && platform->driver->ops->trigger) return platform->driver->ops->trigger(substream, cmd); return 0; } -- cgit From 2062b4c5d2d8fa47dc89105464e67a7ba310c9e7 Mon Sep 17 00:00:00 2001 From: Russell King - ARM Linux Date: Thu, 31 Oct 2013 15:09:20 +0000 Subject: ASoC: dpcm: improve robustness Avoid oopsing if there is no backend stream associated with a front end stream. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d81b79251760..591f0f3074c5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1038,6 +1038,12 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); + if (!be_substream) { + dev_err(be->dev, "ASoC: no backend %s stream\n", + stream ? "capture" : "playback"); + continue; + } + /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; @@ -1055,7 +1061,8 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) continue; - dev_dbg(be->dev, "ASoC: open BE %s\n", be->dai_link->name); + dev_dbg(be->dev, "ASoC: open %s BE %s\n", + stream ? "capture" : "playback", be->dai_link->name); be_substream->runtime = be->dpcm[stream].runtime; err = soc_pcm_open(be_substream); -- cgit From 7b5bfb82882b9b1c8423ce0ed6852ca3762d967a Mon Sep 17 00:00:00 2001 From: Phil Edworthy Date: Thu, 31 Oct 2013 23:06:17 -0700 Subject: ASoC: ak4642: prevent un-necessary changes to SG_SL1 If you record the sound during playback, the playback sound becomes silent. Modify so that the codec driver does not clear SG_SL1::DACL bit which is controlled under widget Signed-off-by: Phil Edworthy Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d0378709702..687565d08d9c 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); -- cgit From 023838542dc8a4eac9650f98942671078a4ce73d Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 31 Oct 2013 18:31:51 -0400 Subject: ALSA: hda - not choose assigned converters for unused pins of Valleyview For Valleyview display codec, if an unused pin chooses an assgined converter selected by a used pin, playback on the unused pin can also give sound to the output device of the used pin. It's because data flows from the same convertor to the display port of the used pin. This issue is same as Haswell. So this patch avoids using assinged convertors for unused pins. The related function haswell_config_cvts() is renamed for code reuse. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 772827b9fb20..d61cc2ab52af 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -46,6 +46,7 @@ module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) +#define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1327,7 +1328,7 @@ static int hdmi_choose_cvt(struct hda_codec *codec, return 0; } -static void haswell_config_cvts(struct hda_codec *codec, +static void not_share_unassigned_cvt(struct hda_codec *codec, hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; @@ -1406,8 +1407,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell(codec)) - haswell_config_cvts(codec, per_pin->pin_nid, mux_idx); + if (is_haswell(codec) || is_valleyview(codec)) + not_share_unassigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); -- cgit From 43bc3bf64b30cdfdffdc41e33bf21222e9396c42 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 1 Nov 2013 15:56:52 +0000 Subject: ASoC: wm_adsp: Print error when regmap reads/writes fail Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b38f3506418f..076da025ba84 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -396,8 +396,8 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to write %zu bytes to %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to write %zu bytes to %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -450,8 +450,8 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to read %zu bytes from %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to read %zu bytes from %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -1313,8 +1313,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write to %x in %s\n", - file, blocks, reg, region_name); + "%s.%d: Failed to write to %x in %s: %d\n", + file, blocks, reg, region_name, ret); } } -- cgit From 7328823d0052bbdb15af162f9f510ced811bdfe8 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 1 Nov 2013 15:56:53 +0000 Subject: ASoC: wm_adsp: Release firmware on memory allocation failure Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 076da025ba84..4008ceb77c5b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -689,7 +689,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } ret = regmap_raw_write_async(regmap, reg, buf->buf, -- cgit From 562c5e6f52bc9ed48b8dc9cef97923b64bd843ec Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 1 Nov 2013 15:56:55 +0000 Subject: ASoC: wm_adsp: Add debug info on get()/put() transfers Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4008ceb77c5b..1f1fc0dd716e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -401,6 +401,7 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, kfree(scratch); return ret; } + adsp_dbg(adsp, "Wrote %zu bytes to %x\n", ctl->len, reg); kfree(scratch); @@ -455,6 +456,7 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, kfree(scratch); return ret; } + adsp_dbg(adsp, "Read %zu bytes from %x\n", ctl->len, reg); memcpy(buf, scratch, ctl->len); kfree(scratch); -- cgit From b0101b4f14d591719f53f7f38ede3651113e6a53 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 1 Nov 2013 15:56:56 +0000 Subject: ASoC: wm_adsp: Remove and free algorithm regions for ADSP1 Do it in a similar fashion as we do for ADSP2. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1f1fc0dd716e..cc3575b5783f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1361,6 +1361,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; int val; @@ -1438,6 +1439,14 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; + + while (!list_empty(&dsp->alg_regions)) { + alg_region = list_first_entry(&dsp->alg_regions, + struct wm_adsp_alg_region, + list); + list_del(&alg_region->list); + kfree(alg_region); + } break; default: -- cgit From 3626992a21610fa19534d392bb0e79cc55a99c9a Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 1 Nov 2013 15:56:57 +0000 Subject: ASoC: wm_adsp: Print out the firmware version Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index cc3575b5783f..53b6033658a6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -570,6 +570,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) file, header->ver); goto out_fw; } + adsp_info(dsp, "Firmware version: %d\n", header->ver); if (header->core != dsp->type) { adsp_err(dsp, "%s: invalid core %d != %d\n", -- cgit From dea0c74ff9e0d85d07b694e261aa7bda2c314ce8 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 1 Nov 2013 10:02:10 +0000 Subject: ASoC: wm8962: Add ALC coefficient support Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7dd79c4efc13..028686fd3eb9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1775,6 +1775,11 @@ WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1), WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30), + +SOC_DOUBLE("ALC Switch", WM8962_ALC1, WM8962_ALCL_ENA_SHIFT, + WM8962_ALCR_ENA_SHIFT, 1, 0), +SND_SOC_BYTES_MASK("ALC Coefficients", WM8962_ALC1, 4, + WM8962_ALCL_ENA_MASK | WM8962_ALCR_ENA_MASK), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { -- cgit From ae2ff9f6c529ba28adea906037a93fd14e46e052 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 1 Nov 2013 10:02:58 +0000 Subject: ASoC: wm8962: Add EQ coefficient support Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 028686fd3eb9..3a2f96c5442c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1758,6 +1758,9 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv), SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), +SND_SOC_BYTES("EQL Coefficients", WM8962_EQ4, 18), +SND_SOC_BYTES("EQR Coefficients", WM8962_EQ24, 18), + SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0), SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA), -- cgit From 611885bc963a2fa6fbd6141d149abbcce4dd8923 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sun, 3 Nov 2013 17:15:00 +0200 Subject: ALSA: hda - hdmi: Disallow unsupported 2ch remapping on NVIDIA codecs NVIDIA HDMI codecs do not seem to follow the Audio Sample Packet (ASP) channel mapping (as set by verb F32h per HDA specification 7.3.3.41) when playing back 2-channel audio (CEA CA 0x00). Basically this means that specifying swapped channels for stereo audio (FR,FL) does not take effect, and e.g. this command plays back on the wrong channel: speaker-test -c2 -Dhdmi:CARD=NVidia,DEV=0 -m FR,FL -s1 Multichannel audio is not affected. This issue has been confirmed to exist on codec 0x10de0015 by me and on 0x10de0040 by Juho Teperi. Disable 2ch FL/FR channel swapping on all NVIDIA HDMI codecs that use the standard HDA channel mapping system. Since this is a very minor functionality loss, we err on the side of disabling it for newer codecs as well until any future testing confirms that this issue has been fixed. Signed-off-by: Anssi Hannula Helped-by: Juho Teperi Cc: Aaron Plattner Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 86 +++++++++++++++++++++++++++++++++------------- 1 file changed, 63 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d61cc2ab52af..c3e3537b3196 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2786,6 +2786,46 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) return 0; } +/* + * NVIDIA codecs ignore ASP mapping for 2ch - confirmed on: + * - 0x10de0015 + * - 0x10de0040 + */ +static int nvhdmi_chmap_cea_alloc_validate_get_type(struct cea_channel_speaker_allocation *cap, + int channels) +{ + if (cap->ca_index == 0x00 && channels == 2) + return SNDRV_CTL_TLVT_CHMAP_FIXED; + + return hdmi_chmap_cea_alloc_validate_get_type(cap, channels); +} + +static int nvhdmi_chmap_validate(int ca, int chs, unsigned char *map) +{ + if (ca == 0x00 && (map[0] != SNDRV_CHMAP_FL || map[1] != SNDRV_CHMAP_FR)) + return -EINVAL; + + return 0; +} + +static int patch_nvhdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + spec = codec->spec; + + spec->ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->ops.chmap_validate = nvhdmi_chmap_validate; + + return 0; +} + /* * ATI/AMD-specific implementations */ @@ -3167,30 +3207,30 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi }, +{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi }, /* 17 is known to be absent */ -{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, -- cgit From 300016b960661b4df63690177b22ba5426ff5706 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 4 Nov 2013 01:13:13 -0500 Subject: ALSA: hda - rename function not_share_unassigned_cvt() The function name not_share_unassigned_cvt() is opposite to what it does. This patch renames it to intel_not_share_assigned_cvt(), and addes comments to explain why some Intel display codecs need this workaround. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c3e3537b3196..e22323f50424 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1328,7 +1328,16 @@ static int hdmi_choose_cvt(struct hda_codec *codec, return 0; } -static void not_share_unassigned_cvt(struct hda_codec *codec, +/* Intel HDMI workaround to fix audio routing issue: + * For some Intel display codecs, pins share the same connection list. + * So a conveter can be selected by multiple pins and playback on any of these + * pins will generate sound on the external display, because audio flows from + * the same converter to the display pipeline. Also muting one pin may make + * other pins have no sound output. + * So this function assures that an assigned converter for a pin is not selected + * by any other pins. + */ +static void intel_not_share_assigned_cvt(struct hda_codec *codec, hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; @@ -1408,7 +1417,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, /* configure unused pins to choose other converters */ if (is_haswell(codec) || is_valleyview(codec)) - not_share_unassigned_cvt(codec, per_pin->pin_nid, mux_idx); + intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); -- cgit From c01422a4a184a183b03fb3046af88d61828f6d56 Mon Sep 17 00:00:00 2001 From: Nariman Poushin Date: Mon, 4 Nov 2013 12:03:44 +0000 Subject: ASoC: wm_adsp: Interpret ADSP memory region lengths as 32 bit words Pad the ADSP word (3 bytes) to 4 bytes in the kernel and calculate lengths based on padded ADSP words instead of treating them as bytes Signed-off-by: Nariman Poushin Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b38f3506418f..60b6b593c407 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1062,6 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", @@ -1079,6 +1080,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", @@ -1108,6 +1110,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", @@ -1125,6 +1128,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", @@ -1142,6 +1146,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", -- cgit From 9c1fc209138a7c1ee439c7b05fde484441c84e7e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 4 Nov 2013 09:51:20 -0200 Subject: ASoC: fsl: fsl_spdif: No need to check the return value of platform_get_resource() When using devm_ioremap_resource(), we do not need to check the return value of platform_get_resource(), so just remove it. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e1bf5ef31bdd..bab95d7ad0ac 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1107,11 +1107,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "could not determine device resources\n"); - return -ENXIO; - } - regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(regs)) return PTR_ERR(regs); -- cgit From 988e8cc41efecbab53f13184940aa6317047765b Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 4 Nov 2013 14:57:31 +0800 Subject: ASoC: Add pinctrl PM to components of active DAIs It's quite popular that more drivers are using pinctrl PM, for example: (Documentation/devicetree/bindings/arm/primecell.txt). Just like what runtime PM does, it would deactivate and activate pin group depending on whether it's being used or not. And this pinctrl PM might be also beneficial to cpu dai drivers because they might have actual pinctrl so as to sleep their pins and wake them up as needed. To achieve this goal, this patch sets pins to the default state during resume or startup; While during suspend and shutdown, it would set pins to the sleep state. As pinctrl PM would return zero if there is no such pinctrl sleep state settings, this patch would not break current ASoC subsystem directly. [ However, there is still an exception that the patch can not handle, that is, when cpu dai driver does not have pinctrl property but another device has it. (The AUDMUX <-> SSI on Freescale i.MX6 series for example. SSI as a cpu dai doesn't contain pinctrl property while AUDMUX, an Audio Multiplexer, has it). In this case, this kind of cpu dai driver needs to find a way to obtain the pinctrl property as its own, by moving property from AUDMUX to SSI, or creating a pins link/dependency between these two devices, or using a more decent way after we figure it out. ] Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 11 +++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b38e0ee622df..1a4e97773c71 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -662,6 +662,8 @@ int snd_soc_suspend(struct device *dev) codec->cache_sync = 1; if (codec->using_regmap) regcache_mark_dirty(codec->control_data); + /* deactivate pins to sleep state */ + pinctrl_pm_select_sleep_state(codec->dev); break; default: dev_dbg(codec->dev, @@ -679,6 +681,9 @@ int snd_soc_suspend(struct device *dev) if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control) cpu_dai->driver->suspend(cpu_dai); + + /* deactivate pins to sleep state */ + pinctrl_pm_select_sleep_state(cpu_dai->dev); } if (card->suspend_post) @@ -807,6 +812,16 @@ int snd_soc_resume(struct device *dev) if (list_empty(&card->codec_dev_list)) return 0; + /* activate pins from sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + if (cpu_dai->active) + pinctrl_pm_select_default_state(cpu_dai->dev); + if (codec_dai->active) + pinctrl_pm_select_default_state(codec_dai->dev); + } + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume @@ -1930,6 +1945,14 @@ int snd_soc_poweroff(struct device *dev) snd_soc_dapm_shutdown(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + pinctrl_pm_select_sleep_state(codec_dai->dev); + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_poweroff); @@ -3752,6 +3775,16 @@ int snd_soc_register_card(struct snd_soc_card *card) if (ret != 0) soc_cleanup_card_debugfs(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 591f0f3074c5..42782c01e413 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -183,6 +184,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pinctrl_pm_select_default_state(cpu_dai->dev); + pinctrl_pm_select_default_state(codec_dai->dev); pm_runtime_get_sync(cpu_dai->dev); pm_runtime_get_sync(codec_dai->dev); pm_runtime_get_sync(platform->dev); @@ -317,6 +320,10 @@ out: pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); return ret; } @@ -426,6 +433,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); return 0; } -- cgit From 4eeca499be4ff4216b745e35ae8c8bffa6445eac Mon Sep 17 00:00:00 2001 From: James Ralston Date: Mon, 4 Nov 2013 09:27:45 -0800 Subject: ALSA: hda - Add Device IDs for Intel Wildcat Point-LP PCH This patch adds the HD Audio Device IDs for the Intel Wildcat Point-LP PCH. Signed-off-by: James Ralston Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a0a06f79daa3..66f7e1e3ec4b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -169,6 +169,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, PPT}," "{Intel, LPT}," "{Intel, LPT_LP}," + "{Intel, WPT_LP}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -3986,6 +3987,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c21), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + /* Wildcat Point-LP */ + { PCI_DEVICE(0x8086, 0x9ca0), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | -- cgit From 24eff328f65c8ef352c90b6adb7c2f39eb94205d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2013 18:21:08 +0100 Subject: ALSA: hda - Enable SPDIF for Acer TravelMate 6293 BIOS on Acer TravelMate 6293 doesn't set up the SPDIF output pin correctly as default, so enable it via a fixup entry. Reported-and-tested-by: Hagen Heiduck Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d7c86d81196..01bf812455a2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2386,6 +2386,7 @@ static const struct hda_verb alc268_beep_init_verbs[] = { enum { ALC268_FIXUP_INV_DMIC, ALC268_FIXUP_HP_EAPD, + ALC268_FIXUP_SPDIF, }; static const struct hda_fixup alc268_fixups[] = { @@ -2400,6 +2401,13 @@ static const struct hda_fixup alc268_fixups[] = { {} } }, + [ALC268_FIXUP_SPDIF] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1e, 0x014b1180 }, /* enable SPDIF out */ + {} + } + }, }; static const struct hda_model_fixup alc268_fixup_models[] = { @@ -2409,6 +2417,7 @@ static const struct hda_model_fixup alc268_fixup_models[] = { }; static const struct snd_pci_quirk alc268_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0139, "Acer TravelMate 6293", ALC268_FIXUP_SPDIF), SND_PCI_QUIRK(0x1025, 0x015b, "Acer AOA 150 (ZG5)", ALC268_FIXUP_INV_DMIC), /* below is codec SSID since multiple Toshiba laptops have the * same PCI SSID 1179:ff00 -- cgit From 0dca01c37a68017fe7112b46bc4b48c927db18c6 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 5 Nov 2013 04:41:06 +0100 Subject: ALSA: usb: supply channel maps even when wChannelConfig is unspecified If wChannelconfig is given for some formats but not others, userspace might not be able to set the channel map. This is RFC because I'm not sure what the best behaviour is - to guess the channel map from the given number of channels (it's quite likely that one channel is MONO and two channels is FL FR), or just to supply UNKNOWN for all channels. But the complete lack of channel map for a format leads userspace to believe that the format is not available at all. Or am I misunderstanding how this should be used? Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4339f97226b..b43b6eec3227 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -281,8 +281,6 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, const unsigned int *maps; int c; - if (!bits) - return NULL; if (channels > ARRAY_SIZE(chmap->map)) return NULL; @@ -293,9 +291,19 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, maps = protocol == UAC_VERSION_2 ? uac2_maps : uac1_maps; chmap->channels = channels; c = 0; - for (; bits && *maps; maps++, bits >>= 1) { - if (bits & 1) - chmap->map[c++] = *maps; + + if (bits) { + for (; bits && *maps; maps++, bits >>= 1) + if (bits & 1) + chmap->map[c++] = *maps; + } else { + /* If we're missing wChannelConfig, then guess something + to make sure the channel map is not skipped entirely */ + if (channels == 1) + chmap->map[c++] = SNDRV_CHMAP_MONO; + else + for (; c < channels && *maps; maps++) + chmap->map[c++] = *maps; } for (; c < channels; c++) -- cgit From e3e35f750fff74e701c8913fd7dd714e37a848cd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 5 Nov 2013 04:41:07 +0100 Subject: ALSA: usb - For class 2 devices, use channel map from altsettings The channel config from the streaming descriptor is probably a better indicator of the channel map than the input terminal. Use the input terminal's channel map as fallback only. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index b43b6eec3227..badd1d6d175d 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -587,6 +587,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) num_channels = as->bNrChannels; format = le32_to_cpu(as->bmFormats); + chconfig = le32_to_cpu(as->bmChannelConfig); /* lookup the terminal associated to this interface * to extract the clock */ @@ -594,7 +595,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) as->bTerminalLink); if (input_term) { clock = input_term->bCSourceID; - chconfig = le32_to_cpu(input_term->bmChannelConfig); + if (!chconfig && (num_channels == input_term->bNrChannels)) + chconfig = le32_to_cpu(input_term->bmChannelConfig); break; } -- cgit From 504333df8ba5fc310260285a22ab5d7cf3208795 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 5 Nov 2013 04:41:08 +0100 Subject: ALSA: usb - Don't trust the channel config if the channel count changed In case the channel count of the input terminal is not the same as the channel count of the streaming descriptor, the channel config of the input terminal can not be trusted. Instead fall back to a default (guessed) channel map. This was found on a Logitech USB Headset. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index badd1d6d175d..d737d0e6e558 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -662,7 +662,6 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) * (fp->maxpacksize & 0x7ff); fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); fp->clock = clock; - fp->chmap = convert_chmap(num_channels, chconfig, protocol); /* some quirks for attributes here */ @@ -698,12 +697,16 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) /* ok, let's parse further... */ if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream) < 0) { kfree(fp->rate_table); - kfree(fp->chmap); kfree(fp); fp = NULL; continue; } + /* Create chmap */ + if (fp->channels != num_channels) + chconfig = 0; + fp->chmap = convert_chmap(fp->channels, chconfig, protocol); + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { -- cgit From a4e7a121685a137eeeb01f05d5ed570c1b45017a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2013 15:44:09 +0100 Subject: ALSA: hda - Fix possible zero-division Check the TLV db scale result before actually dividing in vmaster slave init code. Also mask TLV_DB_SCALE_MUTE bit so that the right value is obtained even if this bit is set by the codec driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index de1a7670ba0d..33c01d551aa4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2676,8 +2676,13 @@ static int get_kctl_0dB_offset(struct snd_kcontrol *kctl) set_fs(fs); } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) tlv = kctl->tlv.p; - if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) - val = -tlv[2] / tlv[3]; + if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) { + int step = tlv[3]; + step &= ~TLV_DB_SCALE_MUTE; + if (!step) + return -1; + val = -tlv[2] / step; + } return val; } -- cgit From 485e3e0cdf353e2e37570eafaad7208138bb6620 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2013 15:51:00 +0100 Subject: ALSA: hda - Add sanity check of vmaster slave dB steps Check whether all vmaster slaves have the same dB steps. Otherwise the behavior would become inconsistent. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 33c01d551aa4..dd5403d40830 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2661,7 +2661,7 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) } /* guess the value corresponding to 0dB */ -static int get_kctl_0dB_offset(struct snd_kcontrol *kctl) +static int get_kctl_0dB_offset(struct snd_kcontrol *kctl, int *step_to_check) { int _tlv[4]; const int *tlv = NULL; @@ -2681,6 +2681,12 @@ static int get_kctl_0dB_offset(struct snd_kcontrol *kctl) step &= ~TLV_DB_SCALE_MUTE; if (!step) return -1; + if (*step_to_check && *step_to_check != step) { + snd_printk(KERN_ERR "hda_codec: Mismatching dB step for vmaster slave (%d!=%d)\n", + *step_to_check, step); + return -1; + } + *step_to_check = step; val = -tlv[2] / step; } return val; @@ -2703,7 +2709,7 @@ static int put_kctl_with_value(struct snd_kcontrol *kctl, int val) /* initialize the slave volume with 0dB */ static int init_slave_0dB(void *data, struct snd_kcontrol *slave) { - int offset = get_kctl_0dB_offset(slave); + int offset = get_kctl_0dB_offset(slave, data); if (offset > 0) put_kctl_with_value(slave, offset); return 0; @@ -2764,9 +2770,11 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, /* init with master mute & zero volume */ put_kctl_with_value(kctl, 0); - if (init_slave_vol) + if (init_slave_vol) { + int step = 0; map_slaves(codec, slaves, suffix, - tlv ? init_slave_0dB : init_slave_unmute, kctl); + tlv ? init_slave_0dB : init_slave_unmute, &step); + } if (ctl_ret) *ctl_ret = kctl; -- cgit From a1114a8c681b0724d6ad905f53ff06aa756f5fb8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2013 16:32:01 +0100 Subject: ALSA: hda - Introduce the bitmask for excluding output volume Add a bitmask to hda_gen_spec indicating NIDs to exclude from the possible volume controls. That is, when the bit is set, the NID corresponding to the bit won't be picked as an output volume control any longer. Basically this is just a band-aid for working around the issue found with CS4208 codec, where only the headphone pin has a volume AMP with different dB steps. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60811 Cc: [v3.12+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 8 ++++++-- sound/pci/hda/hda_generic.h | 3 +++ sound/pci/hda/patch_cirrus.c | 2 ++ 3 files changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b7c89dff7066..276f6e759bac 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -549,11 +549,15 @@ static hda_nid_t look_for_out_mute_nid(struct hda_codec *codec, static hda_nid_t look_for_out_vol_nid(struct hda_codec *codec, struct nid_path *path) { + struct hda_gen_spec *spec = codec->spec; int i; for (i = path->depth - 1; i >= 0; i--) { - if (nid_has_volume(codec, path->path[i], HDA_OUTPUT)) - return path->path[i]; + hda_nid_t nid = path->path[i]; + if ((spec->out_vol_mask >> nid) & 1) + continue; + if (nid_has_volume(codec, nid, HDA_OUTPUT)) + return nid; } return 0; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 48d44026705b..7e45cb44d151 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -242,6 +242,9 @@ struct hda_gen_spec { /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; + /* bitmask for skipping volume controls */ + u64 out_vol_mask; + /* badness tables for output path evaluations */ const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 18d972501585..38d073ed4bcc 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -660,6 +660,8 @@ static int patch_cs4208(struct hda_codec *codec) return -ENOMEM; spec->gen.automute_hook = cs_automute; + /* exclude NID 0x10 (HP) from output volumes due to different steps */ + spec->gen.out_vol_mask = 1ULL << 0x10; snd_hda_pick_fixup(codec, cs4208_models, cs4208_fixup_tbl, cs4208_fixups); -- cgit From bbaa0d6665bc14133d7eb573d2b5ff898a06f365 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 5 Nov 2013 09:27:10 +0100 Subject: ALSA: hda: add device IDs for AMD Evergreen/Northern Islands HDMI The device IDs of the AMD Cypress/Juniper/Redwood/Cedar/Cayman/Antilles/ Barts/Turks/Caicos HDMI HDA controllers weren't added explicitly because the generic entry works, but it made the device appearing as "Generic", and people are confused as if it's no proper HDMI controller. Add them so that the name shows up properly as "ATI HDMI" instead of "Generic". According to Takashi's tests and the lack of complaints, these devices work fine without disabling snooping. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 66f7e1e3ec4b..d6dcec768594 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -4079,6 +4079,22 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0xaa48), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa50), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa58), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa60), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa68), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa80), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa88), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa90), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaa98), + .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x9902), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0xaaa0), -- cgit From b67ae3f1c97e2e8532ce051f0ed4365e9d796590 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 5 Nov 2013 13:11:37 +0100 Subject: ALSA: hda - Enable Thinkpad mute/micmute LEDs for Realtek Same as we already have for Conexant. Right now it's only enabled for one machine. Tested-by: Hui Wang Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 83 +++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 81 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01bf812455a2..02071a58f3f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -829,7 +829,11 @@ static inline void alc_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } -#define alc_free snd_hda_gen_free +static void alc_free(struct hda_codec *codec) +{ + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_FREE); + snd_hda_gen_free(codec); +} #ifdef CONFIG_PM static void alc_power_eapd(struct hda_codec *codec) @@ -3527,6 +3531,74 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + +#include + +static int (*led_set_func)(int, bool); + +static void update_tpacpi_mute_led(void *private_data, int enabled) +{ + if (led_set_func) + led_set_func(TPACPI_LED_MUTE, !enabled); +} + +static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + if (!ucontrol || !led_set_func) + return; + if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { + /* TODO: How do I verify if it's a mono or stereo here? */ + bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; + led_set_func(TPACPI_LED_MICMUTE, !val); + } +} + +static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!led_set_func) + led_set_func = symbol_request(tpacpi_led_set); + if (!led_set_func) { + snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); + return; + } + + removefunc = true; + if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + spec->gen.vmaster_mute.hook = update_tpacpi_mute_led; + removefunc = false; + } + if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { + if (spec->gen.num_adc_nids > 1) + snd_printdd("Skipping micmute LED control due to several ADCs"); + else { + spec->gen.cap_sync_hook = update_tpacpi_micmute_led; + removefunc = false; + } + } + } + + if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(tpacpi_led_set); + led_set_func = NULL; + } +} + +#else + +static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3570,6 +3642,7 @@ enum { ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, ALC290_FIXUP_MONO_SPEAKERS, + ALC269_FIXUP_THINKPAD_ACPI, }; static const struct hda_fixup alc269_fixups[] = { @@ -3867,6 +3940,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, }, + [ALC269_FIXUP_THINKPAD_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_thinkpad_acpi, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3950,7 +4029,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), -- cgit From a1c22cdc77744aa2f72504c42fccadef739d4159 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Tue, 5 Nov 2013 10:29:49 +0100 Subject: ALSA: atmel: remove dependency on This include is completely unused since the AT91 sound driver actually uses gpiolib properly. Signed-off-by: Linus Walleij Acked-by: Nicolas Ferre Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index ae63d22c0f88..c5f0ddd729b3 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,7 +34,6 @@ #include #include -#include #ifdef CONFIG_ARCH_AT91 #include -- cgit From 33499a15c2f7addc81695778753c2338b960eff7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 17:34:46 +0100 Subject: ALSA: hda - Force buffer alignment for Haswell HDMI controllers Haswell HDMI audio controllers seem to get stuck when unaligned buffer size is used. Let's enable the buffer alignment for the corresponding entries. Since AZX_DCAPS_INTEL_PCH contains AZX_DCAPS_BUFSIZE that disables the buffer alignment forcibly, define AZX_DCAPS_INTEL_HASWELL and put the necessary AZX_DCAPS bits there. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60769 Reported-by: Alexander E. Patrakov Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d6dcec768594..9cbd1259a5be 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -613,6 +613,11 @@ enum { #define AZX_DCAPS_INTEL_PCH \ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) +#define AZX_DCAPS_INTEL_HASWELL \ + (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \ + AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME | \ + AZX_DCAPS_I915_POWERWELL) + /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ (AZX_DCAPS_ATI_SNOOP | AZX_DCAPS_NO_TCSEL | \ @@ -3992,14 +3997,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | - AZX_DCAPS_I915_POWERWELL }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, { PCI_DEVICE(0x8086, 0x0c0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | - AZX_DCAPS_I915_POWERWELL }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, { PCI_DEVICE(0x8086, 0x0d0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | - AZX_DCAPS_I915_POWERWELL }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, -- cgit From fab1285a51b7bf55adb4678d82e606829c9dab85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 17:54:05 +0100 Subject: ALSA: hda - Name Haswell HDMI controllers better "HDA Intel MID" is no correct name for Haswell HDMI controllers. Give them a better name, "HDA Intel HDMI". Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9cbd1259a5be..17f4aa8cddd5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -569,6 +569,7 @@ enum { AZX_DRIVER_ICH, AZX_DRIVER_PCH, AZX_DRIVER_SCH, + AZX_DRIVER_HDMI, AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, @@ -648,6 +649,7 @@ static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", [AZX_DRIVER_SCH] = "HDA Intel MID", + [AZX_DRIVER_HDMI] = "HDA Intel HDMI", [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", @@ -3997,11 +3999,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, + .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, { PCI_DEVICE(0x8086, 0x0c0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, + .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, { PCI_DEVICE(0x8086, 0x0d0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, + .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, -- cgit From 8fe7b65ab4656e5db466a7d98b1fd48ff83b2c64 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 10:31:07 +0100 Subject: ALSA: hda - Apply GPIO setup for MacBooks with CS4208 Apply the existing GPIO0 fixup as default for MacBooks with CS4208 codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64401 Cc: [v3.12+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 38d073ed4bcc..1ce1f4067b15 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -611,6 +611,7 @@ static const struct snd_pci_quirk cs4208_fixup_tbl[] = { /* codec SSID */ SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6), SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6), + SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS4208_GPIO0), {} /* terminator */ }; -- cgit From 791b3f596e2331f80244cc8532a25b48d45f126d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 08:09:35 +0100 Subject: ALSA: intel8x0: Fix chmap application The playback chmap for multi-channel stream hasn't been properly added to intel8x0 devices due to the wrong condition. Reported-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 59c8aaebb91e..08d8733604a2 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1541,17 +1541,16 @@ static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device, snd_dma_pci_data(chip->pci), rec->prealloc_size, rec->prealloc_max_size); - if (rec->ac97_idx == ICHD_PCMOUT && rec->playback_ops) { + if (rec->playback_ops && + rec->playback_ops->open == snd_intel8x0_playback_open) { struct snd_pcm_chmap *chmap; int chs = 2; - if (rec->ac97_idx == ICHD_PCMOUT) { - if (chip->multi8) - chs = 8; - else if (chip->multi6) - chs = 6; - else if (chip->multi4) - chs = 4; - } + if (chip->multi8) + chs = 8; + else if (chip->multi6) + chs = 6; + else if (chip->multi4) + chs = 4; err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, snd_pcm_alt_chmaps, chs, 0, &chmap); -- cgit From 29fa9578046ea51e9cf97c4e16207ea6ec375084 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 15:00:02 +0100 Subject: ALSA: ctxfi: Use WARN_ON() instead of BUG_ON() BUG_ON() is rather useless for debugging as it leads to panic(). Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthardware.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c index 110b8ace6d8a..a689f2552706 100644 --- a/sound/pci/ctxfi/cthardware.c +++ b/sound/pci/ctxfi/cthardware.c @@ -69,7 +69,8 @@ unsigned int get_field(unsigned int data, unsigned int field) { int i; - BUG_ON(!field); + if (WARN_ON(!field)) + return 0; /* @field should always be greater than 0 */ for (i = 0; !(field & (1 << i)); ) i++; @@ -81,7 +82,8 @@ void set_field(unsigned int *data, unsigned int field, unsigned int value) { int i; - BUG_ON(!field); + if (WARN_ON(!field)) + return; /* @field should always be greater than 0 */ for (i = 0; !(field & (1 << i)); ) i++; -- cgit From 025be74c1db44158dd3cfe43b89ba16266d01062 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 15:01:47 +0100 Subject: ALSA: ps3: Use WARN_ON() instead of BUG_ON() BUG_ON() is rather useless for debugging as it leads to panic(). Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 8c7dcbe0118d..ebb76f2d90d7 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -933,8 +933,10 @@ static int snd_ps3_driver_probe(struct ps3_system_bus_device *dev) int i, ret; u64 lpar_addr, lpar_size; - BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1)); - BUG_ON(dev->match_id != PS3_MATCH_ID_SOUND); + if (WARN_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1))) + return -ENODEV; + if (WARN_ON(dev->match_id != PS3_MATCH_ID_SOUND)) + return -ENODEV; the_card.ps3_dev = dev; -- cgit From 5a19b178d7b4dd162b3d4c7e729f30348f2eb177 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 15:02:42 +0100 Subject: ALSA: sparc/cs4231: Use WARN_ON() instead of BUG_ON() BUG_ON() is rather useless for debugging as it leads to panic(). Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai --- sound/sparc/cs4231.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 54aaad2a10f5..b47f6fe6277f 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -429,7 +429,8 @@ static void snd_cs4231_advance_dma(struct cs4231_dma_control *dma_cont, unsigned int period_size = snd_pcm_lib_period_bytes(substream); unsigned int offset = period_size * (*periods_sent); - BUG_ON(period_size >= (1 << 24)); + if (WARN_ON(period_size >= (1 << 24))) + return; if (dma_cont->request(dma_cont, runtime->dma_addr + offset, period_size)) @@ -912,7 +913,8 @@ static int snd_cs4231_playback_prepare(struct snd_pcm_substream *substream) chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO); - BUG_ON(runtime->period_size > 0xffff + 1); + if (WARN_ON(runtime->period_size > 0xffff + 1)) + return -EINVAL; chip->p_periods_sent = 0; spin_unlock_irqrestore(&chip->lock, flags); -- cgit From c3cd1badc86c62f64f7e1d7034b078cb7e9b254b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 15:03:05 +0100 Subject: ALSA: mips/ad1843: Use WARN_ON() instead of BUG_ON() BUG_ON() is rather useless for debugging as it leads to panic(). Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai --- sound/mips/ad1843.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c index c624510ec374..586907500ca5 100644 --- a/sound/mips/ad1843.c +++ b/sound/mips/ad1843.c @@ -276,7 +276,7 @@ static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...) if (reg == -1) reg = fp->reg; else - BUG_ON(reg != fp->reg); + WARN_ON(reg != fp->reg); m = ((1 << fp->nbits) - 1) << fp->lo_bit; mask |= m; bits |= (value << fp->lo_bit) & m; -- cgit From 88ec7ae82dff8ca815ff8a31622a2811240b5d0c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 15:33:40 +0100 Subject: ALSA: pxa2xx: Replace BUG() with snd_BUG() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use snd_BUG() instead. Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97-lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 99a466822a7d..66de90ed30ca 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -208,7 +208,7 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) pxa_ac97_warm_pxa3xx(); else #endif - BUG(); + snd_BUG(); while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) mdelay(1); @@ -245,7 +245,7 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) pxa_ac97_cold_pxa3xx(); else #endif - BUG(); + snd_BUG(); while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) mdelay(1); -- cgit From 68f9672b1388f0999d4087b72c3c80cf1c983fa2 Mon Sep 17 00:00:00 2001 From: Oskar Schirmer Date: Tue, 5 Nov 2013 12:13:54 +0000 Subject: ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation Originally snd_hrtimer_callback() used iprtd->period_time for some jiffies based estimation to determine the right moment to call snd_pcm_period_elapsed(). As timer drifts may well be a problem, this was changed in commit b4e82b5b785670b6 to be based on buffer transmission progress, using iprtd->offset and runtime->buffer_size to calculate the amount of data since last period had elapsed. Unfortunately, iprtd->offset counts in bytes, while runtime->buffer_size counts frames, so adding these to find some delta is like comparing apples and oranges, and eventually results in negative delta values every now and then. This is no big harm, because it simply causes snd_pcm_period_elapsed() being called more often than necessary, as negative delta is taken for a large unsigned value by implicit conversion rule. Nonetheless, the calculation is broken, so one would replace the runtime->buffer_size by its equivalent in bytes. But then, there are chances snd_pcm_period_elapsed() is called late, because calculating the moment for the elapsed period into delta is based against the iprtd->last_offset, which is not necessarily the first byte of the period in question, but some random byte which the FIQ handler left us with in r8/r9 by accident. Again, negative impact is low, as there are plenty of periods already prefilled with data, and snd_pcm_period_elapsed() will probably be called latest when the following period is reached. However, the calculation is conceptually broken, and we are best off removing the clever stuff altogether. snd_pcm_period_elapsed() is now simply called once everytime snd_hrtimer_callback() is run, which may not be most accurate, but at least this way we are quite sure we dont miss an end of period. There is not much extra effort wasted by superfluous calls to snd_pcm_period_elapsed(), as the timer frequency closely matches the period size anyway. Signed-off-by: Oskar Schirmer Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-fiq.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 34043c55f2a6..94720a6dd7b6 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -39,8 +39,6 @@ struct imx_pcm_runtime_data { unsigned int period; int periods; unsigned long offset; - unsigned long last_offset; - unsigned long size; struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; @@ -54,7 +52,6 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct snd_pcm_substream *substream = iprtd->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct pt_regs regs; - unsigned long delta; if (!atomic_read(&iprtd->running)) return HRTIMER_NORESTART; @@ -66,19 +63,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) else iprtd->offset = regs.ARM_r9 & 0xffff; - /* How much data have we transferred since the last period report? */ - if (iprtd->offset >= iprtd->last_offset) - delta = iprtd->offset - iprtd->last_offset; - else - delta = runtime->buffer_size + iprtd->offset - - iprtd->last_offset; - - /* If we've transferred at least a period then report it and - * reset our poll time */ - if (delta >= iprtd->period) { - snd_pcm_period_elapsed(substream); - iprtd->last_offset = iprtd->offset; - } + snd_pcm_period_elapsed(substream); hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); @@ -95,11 +80,9 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params) ; + iprtd->period = params_period_bytes(params); iprtd->offset = 0; - iprtd->last_offset = 0; iprtd->poll_time_ns = 1000000000 / params_rate(params) * params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); -- cgit From d5b6b65e75ce607c2734227524e11574317a1c1a Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 6 Nov 2013 10:50:44 +0100 Subject: ALSA: hda - Make sure mute LEDs stay on during runtime suspend (Realtek) Some HP machines with Realtek codecs have mute LEDs connected to VREF pins. However when these go into runtime suspend, the pin powers down and its pin control is disabled, thus disabling the LED too. This patch fixes that issue by making sure that the pin stays in D0 with correct pin control. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/1248465 Tested-by: Franz Hsieh Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 02071a58f3f8..f19cd4fcb024 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2959,6 +2959,23 @@ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval); } +/* Make sure the led works even in runtime suspend */ +static unsigned int led_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + struct alc_spec *spec = codec->spec; + + if (power_state != AC_PWRST_D3 || nid != spec->mute_led_nid) + return power_state; + + /* Set pin ctl again, it might have just been set to 0 */ + snd_hda_set_pin_ctl(codec, nid, + snd_hda_codec_get_pin_target(codec, nid)); + + return AC_PWRST_D0; +} + static void alc269_fixup_hp_mute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2978,6 +2995,7 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, spec->mute_led_nid = pin - 0x0a + 0x18; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; snd_printd("Detected mute LED for %x:%d\n", spec->mute_led_nid, spec->mute_led_polarity); break; @@ -2993,6 +3011,7 @@ static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, spec->mute_led_nid = 0x18; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; } } @@ -3005,6 +3024,7 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, spec->mute_led_nid = 0x19; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; } } -- cgit From ddb146da23cb08851fa418481dd84412f99eec1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:48 +0100 Subject: ASoC: blackfin: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-sport.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 695351241db8..9dfa1241ea66 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -179,8 +179,9 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport) struct dmasg *desc, temp_desc; unsigned long flags; - BUG_ON(sport->dummy_rx_desc == NULL); - BUG_ON(sport->curr_rx_desc == sport->dummy_rx_desc); + if (WARN_ON(!sport->dummy_rx_desc) || + WARN_ON(sport->curr_rx_desc == sport->dummy_rx_desc)) + return -EINVAL; /* Maybe the dummy buffer descriptor ring is damaged */ sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc + 1; @@ -250,8 +251,9 @@ int sport_rx_start(struct sport_device *sport) return -EBUSY; if (sport->tx_run) { /* tx is running, rx is not running */ - BUG_ON(sport->dma_rx_desc == NULL); - BUG_ON(sport->curr_rx_desc != sport->dummy_rx_desc); + if (WARN_ON(!sport->dma_rx_desc) || + WARN_ON(sport->curr_rx_desc != sport->dummy_rx_desc)) + return -EINVAL; local_irq_save(flags); while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - sizeof(struct dmasg)) != sport->dummy_rx_desc) @@ -298,8 +300,9 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) struct dmasg *desc, temp_desc; unsigned long flags; - BUG_ON(sport->dummy_tx_desc == NULL); - BUG_ON(sport->curr_tx_desc == sport->dummy_tx_desc); + if (WARN_ON(!sport->dummy_tx_desc) || + WARN_ON(sport->curr_tx_desc == sport->dummy_tx_desc)) + return -EINVAL; sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc + 1; @@ -331,8 +334,9 @@ int sport_tx_start(struct sport_device *sport) if (sport->tx_run) return -EBUSY; if (sport->rx_run) { - BUG_ON(sport->dma_tx_desc == NULL); - BUG_ON(sport->curr_tx_desc != sport->dummy_tx_desc); + if (WARN_ON(!sport->dma_tx_desc) || + WARN_ON(sport->curr_tx_desc != sport->dummy_tx_desc)) + return -EINVAL; /* Hook the normal buffer descriptor */ local_irq_save(flags); while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - @@ -767,7 +771,8 @@ static irqreturn_t err_handler(int irq, void *dev_id) int sport_set_rx_callback(struct sport_device *sport, void (*rx_callback)(void *), void *rx_data) { - BUG_ON(rx_callback == NULL); + if (WARN_ON(!rx_callback)) + return -EINVAL; sport->rx_callback = rx_callback; sport->rx_data = rx_data; @@ -778,7 +783,8 @@ EXPORT_SYMBOL(sport_set_rx_callback); int sport_set_tx_callback(struct sport_device *sport, void (*tx_callback)(void *), void *tx_data) { - BUG_ON(tx_callback == NULL); + if (WARN_ON(!tx_callback)) + return -EINVAL; sport->tx_callback = tx_callback; sport->tx_data = tx_data; @@ -789,7 +795,8 @@ EXPORT_SYMBOL(sport_set_tx_callback); int sport_set_err_callback(struct sport_device *sport, void (*err_callback)(void *), void *err_data) { - BUG_ON(err_callback == NULL); + if (WARN_ON(!err_callback)) + return -EINVAL; sport->err_callback = err_callback; sport->err_data = err_data; @@ -856,7 +863,8 @@ struct sport_device *sport_init(struct platform_device *pdev, param.wdsize = wdsize; param.dummy_count = dummy_count; - BUG_ON(param.wdsize == 0 || param.dummy_count == 0); + if (WARN_ON(param.wdsize == 0 || param.dummy_count == 0)) + return NULL; ret = sport_config_pdev(pdev, ¶m); if (ret) -- cgit From bee026d0fe4bcc2b29e9279c319ed5e60a3bda97 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:49 +0100 Subject: ASoC: max98088: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 566a367c94fa..8bd2d8a6a2f5 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -621,8 +621,9 @@ static void m98088_eq_band(struct snd_soc_codec *codec, unsigned int dai, unsigned int eq_reg; unsigned int i; - BUG_ON(band > 4); - BUG_ON(dai > 1); + if (WARN_ON(band > 4) || + WARN_ON(dai > 1)) + return; /* Load the base register address */ eq_reg = dai ? M98088_REG_84_DAI2_EQ_BASE : M98088_REG_52_DAI1_EQ_BASE; @@ -962,7 +963,8 @@ static int max98088_line_pga(struct snd_soc_dapm_widget *w, struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); u8 *state; - BUG_ON(!((channel == 1) || (channel == 2))); + if (WARN_ON(!(channel == 1 || channel == 2))) + return -EINVAL; switch (line) { case LINE_INA: -- cgit From a922cd7151371f429c7456951b77999c75520955 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:50 +0100 Subject: ASoC: max98095: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 8dbcacd44e6a..04618a5f2a6f 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -637,8 +637,9 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, unsigned int eq_reg; unsigned int i; - BUG_ON(band > 4); - BUG_ON(dai > 1); + if (WARN_ON(band > 4) || + WARN_ON(dai > 1)) + return; /* Load the base register address */ eq_reg = dai ? M98095_142_DAI2_EQ_BASE : M98095_110_DAI1_EQ_BASE; @@ -662,8 +663,9 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, unsigned int bq_reg; unsigned int i; - BUG_ON(band > 1); - BUG_ON(dai > 1); + if (WARN_ON(band > 1) || + WARN_ON(dai > 1)) + return; /* Load the base register address */ bq_reg = dai ? M98095_17E_DAI2_BQ_BASE : M98095_174_DAI1_BQ_BASE; @@ -1011,7 +1013,8 @@ static int max98095_line_pga(struct snd_soc_dapm_widget *w, struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); u8 *state; - BUG_ON(!((channel == 1) || (channel == 2))); + if (WARN_ON(!(channel == 1 || channel == 2))) + return -EINVAL; state = &max98095->lin_state; @@ -1868,7 +1871,8 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, int fs, best, best_val, i; int regmask, regsave; - BUG_ON(channel > 1); + if (WARN_ON(channel > 1)) + return -EINVAL; if (!pdata || !max98095->eq_textcnt) return 0; -- cgit From 773392b25ce1e5d99c72fcc227bb53ba358497be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:51 +0100 Subject: ASoC: tpa6130a2: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index c58bee8346ce..348552e1771e 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -55,7 +55,8 @@ static int tpa6130a2_i2c_read(int reg) struct tpa6130a2_data *data; int val; - BUG_ON(tpa6130a2_client == NULL); + if (WARN_ON(!tpa6130a2_client)) + return -EINVAL; data = i2c_get_clientdata(tpa6130a2_client); /* If powered off, return the cached value */ @@ -77,7 +78,8 @@ static int tpa6130a2_i2c_write(int reg, u8 value) struct tpa6130a2_data *data; int val = 0; - BUG_ON(tpa6130a2_client == NULL); + if (WARN_ON(!tpa6130a2_client)) + return -EINVAL; data = i2c_get_clientdata(tpa6130a2_client); if (data->power_state) { @@ -98,7 +100,8 @@ static u8 tpa6130a2_read(int reg) { struct tpa6130a2_data *data; - BUG_ON(tpa6130a2_client == NULL); + if (WARN_ON(!tpa6130a2_client)) + return 0; data = i2c_get_clientdata(tpa6130a2_client); return data->regs[reg]; @@ -109,7 +112,8 @@ static int tpa6130a2_initialize(void) struct tpa6130a2_data *data; int i, ret = 0; - BUG_ON(tpa6130a2_client == NULL); + if (WARN_ON(!tpa6130a2_client)) + return -EINVAL; data = i2c_get_clientdata(tpa6130a2_client); for (i = 1; i < TPA6130A2_REG_VERSION; i++) { @@ -127,7 +131,8 @@ static int tpa6130a2_power(u8 power) u8 val; int ret = 0; - BUG_ON(tpa6130a2_client == NULL); + if (WARN_ON(!tpa6130a2_client)) + return -EINVAL; data = i2c_get_clientdata(tpa6130a2_client); mutex_lock(&data->mutex); @@ -193,7 +198,8 @@ static int tpa6130a2_get_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - BUG_ON(tpa6130a2_client == NULL); + if (WARN_ON(!tpa6130a2_client)) + return -EINVAL; data = i2c_get_clientdata(tpa6130a2_client); mutex_lock(&data->mutex); @@ -223,7 +229,8 @@ static int tpa6130a2_put_volsw(struct snd_kcontrol *kcontrol, unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val_reg; - BUG_ON(tpa6130a2_client == NULL); + if (WARN_ON(!tpa6130a2_client)) + return -EINVAL; data = i2c_get_clientdata(tpa6130a2_client); if (invert) -- cgit From f5b3a563943f415b99823be9918119f07337e328 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:52 +0100 Subject: ASoC: wm0010: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index d5ebcb00019b..aea916431227 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -372,7 +372,8 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) offset = 0; dsp = inforec->dsp_target; wm0010->boot_failed = false; - BUG_ON(!list_empty(&xfer_list)); + if (WARN_ON(!list_empty(&xfer_list))) + return -EINVAL; init_completion(&done); /* First record should be INFO */ -- cgit From bf90e895b51f61722681c1ee6e99113cbeba7d13 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:53 +0100 Subject: ASoC: wm2000: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 7fefd766b582..8ae50274ea8f 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -137,7 +137,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) unsigned long rate; int ret; - BUG_ON(wm2000->anc_mode != ANC_OFF); + if (WARN_ON(wm2000->anc_mode != ANC_OFF)) + return -EINVAL; dev_dbg(&i2c->dev, "Beginning power up\n"); @@ -277,7 +278,8 @@ static int wm2000_enter_bypass(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - BUG_ON(wm2000->anc_mode != ANC_ACTIVE); + if (WARN_ON(wm2000->anc_mode != ANC_ACTIVE)) + return -EINVAL; if (analogue) { wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, @@ -315,7 +317,8 @@ static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - BUG_ON(wm2000->anc_mode != ANC_BYPASS); + if (WARN_ON(wm2000->anc_mode != ANC_BYPASS)) + return -EINVAL; wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); @@ -349,7 +352,8 @@ static int wm2000_enter_standby(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - BUG_ON(wm2000->anc_mode != ANC_ACTIVE); + if (WARN_ON(wm2000->anc_mode != ANC_ACTIVE)) + return -EINVAL; if (analogue) { wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, 248 / 4); @@ -392,7 +396,8 @@ static int wm2000_exit_standby(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - BUG_ON(wm2000->anc_mode != ANC_STANDBY); + if (WARN_ON(wm2000->anc_mode != ANC_STANDBY)) + return -EINVAL; wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); -- cgit From 4a9e0f919c44fad6a2c591c6aed519ca3247515b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:54 +0100 Subject: ASoC: wm8580: Use WARN() instead of BUG_ON() Use WARN() instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 5e9c40fa7eb2..08a414b57b1e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -736,7 +736,7 @@ static int wm8580_set_sysclk(struct snd_soc_dai *dai, int clk_id, break; default: - BUG_ON("Unknown DAI driver ID\n"); + WARN(1, "Unknown DAI driver ID\n"); return -EINVAL; } -- cgit From 95ff71e9383fdb6efca11455b8e495af034b7ce9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:55 +0100 Subject: ASoC: wm5100: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index ac1745d030d6..4cf91deabc02 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1972,7 +1972,8 @@ static void wm5100_set_detect_mode(struct wm5100_priv *wm5100, int the_mode) { struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; - BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); + if (WARN_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes))) + return; gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); regmap_update_bits(wm5100->regmap, WM5100_ACCESSORY_DETECT_MODE_1, -- cgit From 22a7038c3950a7a4c4d1656a04c3ca8da8296324 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:56 +0100 Subject: ASoC: wm8776: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index f31017ed1381..942d58e455f3 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -325,7 +325,8 @@ static int wm8776_set_sysclk(struct snd_soc_dai *dai, struct snd_soc_codec *codec = dai->codec; struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); - BUG_ON(dai->driver->id >= ARRAY_SIZE(wm8776->sysclk)); + if (WARN_ON(dai->driver->id >= ARRAY_SIZE(wm8776->sysclk))) + return -EINVAL; wm8776->sysclk[dai->driver->id] = freq; -- cgit From 246e884b82305c5669a892c04e67864f4e7c10d5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:57 +0100 Subject: ASoC: wm8900: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 7c8257c5a17b..de67e74dca0c 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -691,7 +691,8 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, unsigned int K, Ndiv, Nmod, target; unsigned int div; - BUG_ON(!Fout); + if (WARN_ON(!Fout)) + return -EINVAL; /* The FLL must run at 90-100MHz which is then scaled down to * the output value by FLLCLK_DIV. */ @@ -742,8 +743,9 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, /* Move down to proper range now rounding is done */ fll_div->k = K / 10; - BUG_ON(target != Fout * (fll_div->fllclk_div << 2)); - BUG_ON(!K && target != Fref * fll_div->fll_ratio * fll_div->n); + if (WARN_ON(target != Fout * (fll_div->fllclk_div << 2)) || + WARN_ON(!K && target != Fref * fll_div->fll_ratio * fll_div->n)) + return -EINVAL; return 0; } -- cgit From 4c8d620ac9e3aea0a0c2edf36851b59b44bd12f2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:58 +0100 Subject: ASoC: wm8904: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4dfa8dceeabf..c173ab3cd18a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -658,7 +658,8 @@ SOC_SINGLE_TLV("EQ5 Volume", WM8904_EQ6, 0, 24, 0, eq_tlv), static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - BUG_ON(event != SND_SOC_DAPM_POST_PMU); + if (WARN_ON(event != SND_SOC_DAPM_POST_PMU)) + return -EINVAL; /* Maximum startup time */ udelay(500); -- cgit From fb2e3e70193f6757e536dfdecf21be2e6ee308d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:39:59 +0100 Subject: ASoC: wm9713: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a53e175c015a..acea8927905b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -221,7 +221,8 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; u16 status, rate; - BUG_ON(event != SND_SOC_DAPM_PRE_PMD); + if (WARN_ON(event != SND_SOC_DAPM_PRE_PMD)) + return -EINVAL; /* Gracefully shut down the voice interface. */ status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000; -- cgit From a2b4f8a473efd82d634117a057e0ba64443354cf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2013 00:06:45 -0800 Subject: ASoC: rcar: remove un-needed select from Kconfig config RCAR_CLK_ADG is not exist Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 56d8ff6a402d..14011d90d70a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,7 +37,6 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD - select RCAR_CLK_ADG help This option enables R-Car SUR/SCU/SSIU/SSI sound support -- cgit From 8e35cd4ac9967620e03f027f27db18b2e249dcd7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 6 Nov 2013 11:20:01 +0100 Subject: ALSA: HDA - Limit mic boost and add mute LED for an HP machine This machine has a mute LED as well as a noisy internal mic. Hence it needs quirks for both limiting the mic boost as well as enabling the LED. BugLink: https://bugs.launchpad.net/bugs/1248476 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f19cd4fcb024..215db601267d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3657,6 +3657,7 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, @@ -3927,6 +3928,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, }, + [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_HP_MUTE_LED_MIC1, + }, [ALC269VB_FIXUP_ORDISSIMO_EVE2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4014,6 +4021,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- cgit From 77ed16cc6c4d8ff0fe7b18c53e13826a81bd4d06 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 17:54:02 +0100 Subject: ALSA: cmi8328: Fix compile warnings without CONFIG_PM MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Just add an ifdef CONFIG_PM to shut up the warnings: sound/isa/cmi8328.c:129:13: warning: ‘snd_cmi8328_cfg_save’ defined but not used [-Wunused-function] sound/isa/cmi8328.c:136:13: warning: ‘snd_cmi8328_cfg_restore’ defined but not used [-Wunused-function] Signed-off-by: Takashi Iwai --- sound/isa/cmi8328.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index f84f073fc1e8..ab6b2dc043f1 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -126,6 +126,7 @@ static void snd_cmi8328_cfg_write(u16 port, u8 reg, u8 val) outb(val, port + 3); /* yes, value goes to the same port as index */ } +#ifdef CONFIG_PM static void snd_cmi8328_cfg_save(u16 port, u8 cfg[]) { cfg[0] = snd_cmi8328_cfg_read(port, CFG1); @@ -139,6 +140,7 @@ static void snd_cmi8328_cfg_restore(u16 port, u8 cfg[]) snd_cmi8328_cfg_write(port, CFG2, cfg[1]); snd_cmi8328_cfg_write(port, CFG3, cfg[2]); } +#endif /* CONFIG_PM */ static int snd_cmi8328_mixer(struct snd_wss *chip) { -- cgit From 293db84270c2fc37f56030d09c43c7eb7939a7c3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 17:54:55 +0100 Subject: ALSA: lx6464es: Fix pointer cast compile warnings The warnings are really harmless but annoying. Since they are only about debug prints, and it's at most 32bit DMA, let's just cast to unsigned long. sound/pci/lx6464es/lx6464es.c:457:22: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] sound/pci/lx6464es/lx_core.c:1195:21: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.c | 4 ++-- sound/pci/lx6464es/lx_core.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 3230e57f246c..5fcaaa6da4a8 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -453,8 +453,8 @@ static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream) lower_32_bits(buf), upper_32_bits(buf), &buffer_index); - snd_printdd(LXP "starting: buffer index %x on %p (%d bytes)\n", - buffer_index, (void *)buf, period_bytes); + snd_printdd(LXP "starting: buffer index %x on 0x%lx (%d bytes)\n", + buffer_index, (unsigned long)buf, period_bytes); buf += period_bytes; } diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 633c8607d053..626ecad4dae7 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -1191,8 +1191,8 @@ static int lx_interrupt_request_new_buffer(struct lx6464es *chip, unpack_pointer(buf, &buf_lo, &buf_hi); err = lx_buffer_give(chip, 0, is_capture, period_bytes, buf_lo, buf_hi, &buffer_index); - snd_printdd(LXP "interrupt: gave buffer index %x on %p (%d bytes)\n", - buffer_index, (void *)buf, period_bytes); + snd_printdd(LXP "interrupt: gave buffer index %x on 0x%lx (%d bytes)\n", + buffer_index, (unsigned long)buf, period_bytes); lx_stream->frame_pos = next_pos; spin_unlock_irqrestore(&chip->lock, flags); -- cgit From 8f42d7698751a45cd9f7134a5da49bc5b6206179 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 18:47:42 +0100 Subject: ALSA: hda - Add support for CX20952 It's a superset of the existing CX2075x codecs, so we can reuse the existing parser code. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 993b25c17711..c205bb1747fd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3656,6 +3656,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f15115, .name = "CX20757", .patch = patch_conexant_auto }, + { .id = 0x14f151d7, .name = "CX20952", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -3682,6 +3684,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15111"); MODULE_ALIAS("snd-hda-codec-id:14f15113"); MODULE_ALIAS("snd-hda-codec-id:14f15114"); MODULE_ALIAS("snd-hda-codec-id:14f15115"); +MODULE_ALIAS("snd-hda-codec-id:14f151d7"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- cgit From 487a588d09db0d6508261867df208d8bdc718251 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2013 07:29:30 +0100 Subject: ALSA: hda - Add pincfg fixup for ASUS W5A BIOS on ASUS W5A laptop with ALC880 codec doesn't provide any pin configurations, so we have to set up all pins manually. Reported-and-tested-by: nb Cc: [v3.4+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 215db601267d..44f12104e8a6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1045,6 +1045,7 @@ enum { ALC880_FIXUP_UNIWILL, ALC880_FIXUP_UNIWILL_DIG, ALC880_FIXUP_Z71V, + ALC880_FIXUP_ASUS_W5A, ALC880_FIXUP_3ST_BASE, ALC880_FIXUP_3ST, ALC880_FIXUP_3ST_DIG, @@ -1215,6 +1216,26 @@ static const struct hda_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_ASUS_W5A] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* set up the whole pins as BIOS is utterly broken */ + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x411111f0 }, /* N/A */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x90a60160 }, /* mic */ + { 0x19, 0x411111f0 }, /* N/A */ + { 0x1a, 0x411111f0 }, /* N/A */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0xb743111e }, /* SPDIF out */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_GPIO1, + }, [ALC880_FIXUP_3ST_BASE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -1336,6 +1357,7 @@ static const struct hda_fixup alc880_fixups[] = { static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS W5A", ALC880_FIXUP_ASUS_W5A), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), -- cgit From ad8ff99e6beb8708b0bdefd9d5658324e90200f0 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 7 Nov 2013 09:28:59 +0100 Subject: ALSA: hda - Another Dell headset quirk This machine has a multi-function headset jack. BugLink: https://bugs.launchpad.net/bugs/1248856 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 44f12104e8a6..daf720502385 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4796,6 +4796,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), -- cgit From f44f2a5417b2968a8724b352cc0b2545a6bcb1f4 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 7 Nov 2013 10:08:22 +0100 Subject: ALSA: compress: fix drain calls blocking other compress functions (v6) The drain and drain_notify callback were blocked by low level driver until the draining was complete. Due to this being invoked with big fat mutex held, others ops like reading timestamp, calling pause, drop were blocked. So to fix this we add a new snd_compr_drain_notify() API. This would be required to be invoked by low level driver when drain or partial drain has been completed by the DSP. Thus we make the drain and partial_drain callback as non blocking and driver returns immediately after notifying DSP. The waiting is done while releasing the lock so that other ops can go ahead. [ The commit 917f4b5cba78 was wrongly applied from the preliminary patch. This commit corrects to the final version. Sorry for inconvenience! -- tiwai ] Signed-off-by: Vinod Koul CC: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 11 ++++------- sound/core/compress_offload.c | 31 +++++++++++++++++++------------ 2 files changed, 23 insertions(+), 19 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 175ab3237b58..ae6c3b8ed2f5 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -48,8 +48,6 @@ struct snd_compr_ops; * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep - * @wait: drain wait queue - * @drain_wake: condition for drain wake */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -61,8 +59,6 @@ struct snd_compr_runtime { u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; - wait_queue_head_t wait; - unsigned int drain_wake; void *private_data; }; @@ -177,10 +173,11 @@ static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) { - snd_BUG_ON(!stream); + if (snd_BUG_ON(!stream)) + return; - stream->runtime->drain_wake = 1; - wake_up(&stream->runtime->wait); + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + wake_up(&stream->runtime->sleep); } #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 3eb47d0006a7..d9af6387f37c 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -123,7 +123,6 @@ static int snd_compr_open(struct inode *inode, struct file *f) } runtime->state = SNDRV_PCM_STATE_OPEN; init_waitqueue_head(&runtime->sleep); - init_waitqueue_head(&runtime->wait); data->stream.runtime = runtime; f->private_data = (void *)data; mutex_lock(&compr->lock); @@ -681,8 +680,6 @@ static int snd_compr_stop(struct snd_compr_stream *stream) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); if (!retval) { - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - wake_up(&stream->runtime->sleep); snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; @@ -692,6 +689,8 @@ static int snd_compr_stop(struct snd_compr_stream *stream) static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) { + int ret; + /* * We are called with lock held. So drop the lock while we wait for * drain complete notfication from the driver @@ -703,12 +702,24 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) stream->runtime->state = SNDRV_PCM_STATE_DRAINING; mutex_unlock(&stream->device->lock); - wait_event(stream->runtime->wait, stream->runtime->drain_wake); + /* we wait for drain to complete here, drain can return when + * interruption occurred, wait returned error or success. + * For the first two cases we don't do anything different here and + * return after waking up + */ + + ret = wait_event_interruptible(stream->runtime->sleep, + (stream->runtime->state != SNDRV_PCM_STATE_DRAINING)); + if (ret == -ERESTARTSYS) + pr_debug("wait aborted by a signal"); + else if (ret) + pr_debug("wait for drain failed with %d\n", ret); + wake_up(&stream->runtime->sleep); mutex_lock(&stream->device->lock); - return 0; + return ret; } static int snd_compr_drain(struct snd_compr_stream *stream) @@ -719,17 +730,14 @@ static int snd_compr_drain(struct snd_compr_stream *stream) stream->runtime->state == SNDRV_PCM_STATE_SETUP) return -EPERM; - stream->runtime->drain_wake = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); if (retval) { - pr_err("SND_COMPR_TRIGGER_DRAIN failed %d\n", retval); + pr_debug("SND_COMPR_TRIGGER_DRAIN failed %d\n", retval); wake_up(&stream->runtime->sleep); return retval; } - retval = snd_compress_wait_for_drain(stream); - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - return retval; + return snd_compress_wait_for_drain(stream); } static int snd_compr_next_track(struct snd_compr_stream *stream) @@ -764,10 +772,9 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) if (stream->next_track == false) return -EPERM; - stream->runtime->drain_wake = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); if (retval) { - pr_err("Partial drain returned failure\n"); + pr_debug("Partial drain returned failure\n"); wake_up(&stream->runtime->sleep); return retval; } -- cgit From bffbbc0a2ccb9f3a3235ea6c646030e5fc3d771e Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 7 Nov 2013 11:09:54 +0300 Subject: ALSA: sb16 - info leak in snd_sb_csp_ioctl() There is a 2 byte hole after "info.func_nr" so we could leak unitialized stack information to userspace. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/isa/sb/sb16_csp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index c1aa21edcb65..48da2276683d 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -208,6 +208,7 @@ static int snd_sb_csp_ioctl(struct snd_hwdep * hw, struct file *file, unsigned i switch (cmd) { /* get information */ case SNDRV_SB_CSP_IOCTL_INFO: + memset(&info, 0, sizeof(info)); *info.codec_name = *p->codec_name; info.func_nr = p->func_nr; info.acc_format = p->acc_format; -- cgit From 3b098eb486868d57d7f2666d05b86c19a07df71b Mon Sep 17 00:00:00 2001 From: Chen Gang Date: Thu, 7 Nov 2013 11:14:29 +0800 Subject: ALSA: include/uapi/sound/firewire.h: use "_UAPI" instead of "UAPI" When installing, "scripts/headers_install.sh" will strip guard macro' "_UAPI" to prevent from appearing it to users. And also, all another files which need uapi prefix always use "_UAPI", not "UAPI". So use "_UAPI" instead of "UAPI" on the guard macro, and also give a comment for "#endif". Signed-off-by: Chen Gang Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/uapi/sound/firewire.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index e86131ca49e5..59f5961302bf 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -1,5 +1,5 @@ -#ifndef UAPI_SOUND_FIREWIRE_H_INCLUDED -#define UAPI_SOUND_FIREWIRE_H_INCLUDED +#ifndef _UAPI_SOUND_FIREWIRE_H_INCLUDED +#define _UAPI_SOUND_FIREWIRE_H_INCLUDED #include @@ -48,4 +48,4 @@ struct snd_firewire_get_info { * Returns -EBUSY if the driver is already streaming. */ -#endif +#endif /* _UAPI_SOUND_FIREWIRE_H_INCLUDED */ -- cgit From a894bd7fb539d671149fea9420c94c0fbe6baf7a Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Wed, 6 Nov 2013 10:16:20 +0000 Subject: ASoC: generic-dmaengine-pcm: Clear slave_config memory We currently assume that the DMA Slave Config will be fully populated by the platform, however some DMA Engines make decisions based on zero (default) flags such as DMA_SLAVE_BUSWIDTH_UNDEFINED and as this is a static declaration we need to memset it to clear the data area. Signed-off-by: Lee Jones Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ee0790337ec9..6ad4c7a47f5d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -90,6 +90,8 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_slave_config slave_config; int ret; + memset(&slave_config, 0, sizeof(slave_config)); + if (!pcm->config) prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; else -- cgit From efe4710860fa6ed10dd041f13902f0e06c86e8cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2013 13:38:23 +0100 Subject: ALSA: hda - Delay HDMI presence reports while waiting for ELD information There is a small gap between the jack detection unsolicited event and the time the ELD is updated. When user-space queries the HDMI ELD immediately after receiving the notification, it might fail because of this gap. For avoiding such a problem, this patch tries to delay the HDMI jack detect notification until ELD information is fully updated. The workaround is imperfect, but good enough as a starting point. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e22323f50424..ce412d1a15d0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1140,7 +1140,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, * Unsolicited events */ -static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); +static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { @@ -1166,8 +1166,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; - hdmi_present_sense(get_pin(spec, pin_idx), 1); - snd_hda_jack_report_sync(codec); + if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) + snd_hda_jack_report_sync(codec); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -1475,7 +1475,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } -static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) +static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; @@ -1493,6 +1493,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) int present = snd_hda_pin_sense(codec, pin_nid); bool update_eld = false; bool eld_changed = false; + bool ret; mutex_lock(&per_pin->lock); pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); @@ -1559,7 +1560,12 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &per_pin->eld_ctl->id); unlock: + if ((codec->vendor_id & 0xffff0000) == 0x10020000) + ret = true; /* AMD codecs create ELD by itself */ + else + ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid; mutex_unlock(&per_pin->lock); + return ret; } static void hdmi_repoll_eld(struct work_struct *work) @@ -1570,7 +1576,8 @@ static void hdmi_repoll_eld(struct work_struct *work) if (per_pin->repoll_count++ > 6) per_pin->repoll_count = 0; - hdmi_present_sense(per_pin, per_pin->repoll_count); + if (hdmi_present_sense(per_pin, per_pin->repoll_count)) + snd_hda_jack_report_sync(per_pin->codec); } static void intel_haswell_fixup_connect_list(struct hda_codec *codec, -- cgit From 0f568959de1c90be3f9ec51c271a40d7583f5809 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 7 Nov 2013 13:38:24 +0100 Subject: ALSA: hda - Add a block_report flag to jacks If the jack should not be reported to userspace (e g, because it is in some transitional state), one can set this flag. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_jack.h | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 05b3e3e9108f..afe594411a56 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -286,7 +286,7 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) jack = codec->jacktbl.list; for (i = 0; i < codec->jacktbl.used; i++, jack++) if (jack->nid) { - if (!jack->kctl) + if (!jack->kctl || jack->block_report) continue; state = get_jack_plug_state(jack->pin_sense); snd_kctl_jack_report(codec->bus->card, jack->kctl, state); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 379420c44eef..46e1ea83ce3c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -28,6 +28,7 @@ struct hda_jack_tbl { unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ unsigned int phantom_jack:1; /* a fixed, always present port? */ + unsigned int block_report:1; /* in a transitional state - do not report to userspace */ hda_nid_t gating_jack; /* valid when gating jack plugged */ hda_nid_t gated_jack; /* gated is dependent on this jack */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ -- cgit From 464837a7bc0a3495e3490e3bf85099bb2300efbd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 7 Nov 2013 13:38:25 +0100 Subject: ALSA: hda - block HDMI jack reports while repolling This fixes a race condition in case several monitors are being repolled in parallel. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ce412d1a15d0..a96403a828af 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1477,6 +1477,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { + struct hda_jack_tbl *jack; struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; @@ -1564,6 +1565,11 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) ret = true; /* AMD codecs create ELD by itself */ else ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid; + + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + jack->block_report = !ret; + mutex_unlock(&per_pin->lock); return ret; } -- cgit From 512a4cb9e7285e6ef8104da808dc85806321aec8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2013 15:44:11 +0100 Subject: ALSA: ice1724: Fix compile warning with CONFIG_PROC_FS=n Just added a missing ifdef: sound/pci/ice1712/quartet.c:210:14: warning: 'get_binary' defined but not used [-Wunused-function] Signed-off-by: Takashi Iwai --- sound/pci/ice1712/quartet.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index 975e0357bd5a..71c6003ef338 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -203,6 +203,7 @@ static const char * const ext_clock_names[3] = {"IEC958 In", "Word Clock 1xFS", #define AK4620_DEEMVOL_REG 0x03 #define AK4620_SMUTE (1<<7) +#ifdef CONFIG_PROC_FS /* * Conversion from int value to its binary form. Used for debugging. * The output buffer must be allocated prior to calling the function. @@ -227,6 +228,7 @@ static char *get_binary(char *buffer, int value) buffer[pos] = '\0'; return buffer; } +#endif /* CONFIG_PROC_FS */ /* * Initial setup of the conversion array GPIO <-> rate -- cgit From aff747ebbf68c6efc4d091d54058af89b916f82d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2013 16:39:37 +0100 Subject: ALSA: hda - Get rid of AMD HDMI exception in hdmi_present_sense() Since the recent fake ELD patches, we can remove the check for AMD HDMI in hdmi_present_sense() and decide the return value from eld_valid value. Suggested by Anssi Hannula. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a96403a828af..e68792311bb2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1561,10 +1561,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &per_pin->eld_ctl->id); unlock: - if ((codec->vendor_id & 0xffff0000) == 0x10020000) - ret = true; /* AMD codecs create ELD by itself */ - else - ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid; + ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid; jack = snd_hda_jack_tbl_get(codec, pin_nid); if (jack) -- cgit From 656a22a105c8135a515daae3d79ede7b472c18ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:01 +0100 Subject: ASoC: mid-x86: Use WARN_ON() instead of BUG_ON() BUG_ON() is rather useless for debugging as it leads to panic(). Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 392fc0b8f5b8..b6b5eb698d33 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -40,7 +40,8 @@ static DEFINE_MUTEX(sst_lock); int sst_register_dsp(struct sst_device *dev) { - BUG_ON(!dev); + if (WARN_ON(!dev)) + return -EINVAL; if (!try_module_get(dev->dev->driver->owner)) return -ENODEV; mutex_lock(&sst_lock); @@ -59,7 +60,8 @@ EXPORT_SYMBOL_GPL(sst_register_dsp); int sst_unregister_dsp(struct sst_device *dev) { - BUG_ON(!dev); + if (WARN_ON(!dev)) + return -EINVAL; if (dev != sst) return -EINVAL; -- cgit From 96b61bc546519ae90317980294c161c86bcd140e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:02 +0100 Subject: ASoC: omap: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Acked-by: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 5e8d640d314f..6d216cb6c19b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -344,8 +344,11 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) || - (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)); + if (WARN_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) || + (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0))) { + err = -EINVAL; + goto err4; + } gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); -- cgit From 8a2e2c86e979a9b3a53e880be65d5fd12820b932 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:03 +0100 Subject: ASoC: pxa: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index d5340a088858..c0d648d3339f 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -165,7 +165,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct snd_dmaengine_dai_dma_data *dma_data; - BUG_ON(IS_ERR(clk_i2s)); + if (WARN_ON(IS_ERR(clk_i2s))) + return -EINVAL; clk_prepare_enable(clk_i2s); clk_ena = 1; pxa_i2s_wait(); -- cgit From 4a318f1e6c4331633a91ad67b782860c76d8894b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:04 +0100 Subject: ASoC: s6000: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index d0740a762963..5cfaa5464eba 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -90,7 +90,8 @@ static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) return; } - BUG_ON(period_size & 15); + if (WARN_ON(period_size & 15)) + return; s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel), src, dst, period_size); -- cgit From 8b14719bebfe022f3637c323e76c88b2bc061a61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:05 +0100 Subject: ASoC: rcar: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Acked-by: Kuninori Morimoto Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sh/rcar/scu.c | 3 ++- sound/soc/sh/rcar/ssi.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 2df2e9150b89..92e3f51c3a4f 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -192,7 +192,8 @@ static struct rsnd_mod_ops rsnd_scu_ops = { struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id) { - BUG_ON(id < 0 || id >= rsnd_scu_nr(priv)); + if (WARN_ON(id < 0 || id >= rsnd_scu_nr(priv))) + id = 0; return &((struct rsnd_scu *)(priv->scu) + id)->mod; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fae26d3f79d2..fc010d652069 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -619,7 +619,8 @@ struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv, struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) { - BUG_ON(id < 0 || id >= rsnd_ssi_nr(priv)); + if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv))) + id = 0; return &(((struct rsnd_ssiu *)(priv->ssiu))->ssi + id)->mod; } -- cgit From 5f29d4455992ab2edcaa377d4ef61b60240781ac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:06 +0100 Subject: ASoC: sh: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Acked-by: Kuninori Morimoto Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sh/siu_dai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 9dc24ffa892a..d55babee14f8 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -543,7 +543,8 @@ static void siu_dai_shutdown(struct snd_pcm_substream *substream, /* Stop the siu if the other stream is not using it */ if (!port_info->play_cap) { /* during stmread or stmwrite ? */ - BUG_ON(port_info->playback.rw_flg || port_info->capture.rw_flg); + if (WARN_ON(port_info->playback.rw_flg || port_info->capture.rw_flg)) + return; siu_dai_spbstop(port_info); siu_dai_stop(port_info); } -- cgit From cb1b10262f986f865fdbafd0af3327f15f83b8af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:07 +0100 Subject: ASoC: txx9: Use WARN_ON() instead of BUG_ON() Use WARN_ON() and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 45a6428cba8d..fbd077f4de72 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -115,8 +115,8 @@ static void txx9aclc_dma_complete(void *arg) spin_lock_irqsave(&dmadata->dma_lock, flags); if (dmadata->frag_count >= 0) { dmadata->dmacount--; - BUG_ON(dmadata->dmacount < 0); - tasklet_schedule(&dmadata->tasklet); + if (!WARN_ON(dmadata->dmacount < 0)) + tasklet_schedule(&dmadata->tasklet); } spin_unlock_irqrestore(&dmadata->dma_lock, flags); } @@ -181,7 +181,10 @@ static void txx9aclc_dma_tasklet(unsigned long data) spin_unlock_irqrestore(&dmadata->dma_lock, flags); return; } - BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN); + if (WARN_ON(dmadata->dmacount >= NR_DMA_CHAIN)) { + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return; + } while (dmadata->dmacount < NR_DMA_CHAIN) { dmadata->dmacount++; spin_unlock_irqrestore(&dmadata->dma_lock, flags); -- cgit From a361f4525d5e01d7ebe2adafc263e107a34834b2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:12 +0100 Subject: ASoC: wm8350: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Cc: patches@opensource.wolfsonmicro.com Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index af1318ddb062..a183dcf3d5c1 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -274,7 +274,7 @@ static int pga_event(struct snd_soc_dapm_widget *w, break; default: - BUG(); + WARN(1, "Invalid shift %d\n", w->shift); return -1; } -- cgit From d9dea39671ff0066fa031d6469f251447beed14b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:13 +0100 Subject: ASoC: wm8900: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Cc: patches@opensource.wolfsonmicro.com Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index de67e74dca0c..734209e252c3 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -279,7 +279,8 @@ static int wm8900_hp_event(struct snd_soc_dapm_widget *w, break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); + break; } return 0; -- cgit From 8d8bb1ad1ef6e799e85de50f696d2443aa3e84cb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:14 +0100 Subject: ASoC: wm8904: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Cc: patches@opensource.wolfsonmicro.com Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index c173ab3cd18a..3938fb1c203e 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -741,7 +741,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_r = 3; break; default: - BUG(); + WARN(1, "Invalid reg %d\n", reg); return -EINVAL; } -- cgit From a845d59de67d29ac28c39684dfffd252c9d37215 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:15 +0100 Subject: ASoC: wm8958: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Cc: patches@opensource.wolfsonmicro.com Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8958-dsp2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b0710d817a65..b7488f190d2b 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -348,7 +348,7 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) aif = 1; break; default: - BUG(); + WARN(1, "Invalid path %d\n", path); return; } -- cgit From 69134367c34aa35e895f10911ed3587ba270dcb0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:16 +0100 Subject: ASoC: wm8962: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Cc: patches@opensource.wolfsonmicro.com Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11d80f3b6137..22e42e788649 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1845,7 +1845,7 @@ static int cp_event(struct snd_soc_dapm_widget *w, break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); return -EINVAL; } @@ -1937,7 +1937,7 @@ static int hp_event(struct snd_soc_dapm_widget *w, break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); return -EINVAL; } @@ -1966,7 +1966,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, reg = WM8962_SPKOUTL_VOLUME; break; default: - BUG(); + WARN(1, "Invalid shift %d\n", w->shift); return -EINVAL; } @@ -1974,7 +1974,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: return snd_soc_write(codec, reg, snd_soc_read(codec, reg)); default: - BUG(); + WARN(1, "Invalid event %d\n", event); return -EINVAL; } } @@ -1997,7 +1997,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); return -EINVAL; } -- cgit From d8e9a54414b92efd87e32c7320c894e7d0595158 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:17 +0100 Subject: ASoC: wm8996: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Cc: patches@opensource.wolfsonmicro.com Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 46fe83d2b224..d666de03704a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -608,7 +608,7 @@ static int bg_event(struct snd_soc_dapm_widget *w, wm8996_bg_disable(codec); break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); ret = -EINVAL; } @@ -625,7 +625,7 @@ static int cp_event(struct snd_soc_dapm_widget *w, msleep(5); break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); ret = -EINVAL; } @@ -646,7 +646,7 @@ static int rmv_short_event(struct snd_soc_dapm_widget *w, wm8996->hpout_pending |= w->shift; break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); return -EINVAL; } @@ -767,7 +767,7 @@ static int dcs_start(struct snd_soc_dapm_widget *w, wm8996->dcs_pending |= 1 << w->shift; break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); return -EINVAL; } @@ -1656,7 +1656,7 @@ static int wm8996_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) lrclk_rx_reg = WM8996_AIF2_RX_LRCLK_2; break; default: - BUG(); + WARN(1, "Invalid dai id %d\n", dai->id); return -EINVAL; } @@ -1768,7 +1768,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, dsp_shift = WM8996_DSP2_DIV_SHIFT; break; default: - BUG(); + WARN(1, "Invalid dai id %d\n", dai->id); return -EINVAL; } -- cgit From 9a743400a032766a2b79c88233ea3f6430a59b86 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:18 +0100 Subject: ASoC: wm_hubs: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Cc: patches@opensource.wolfsonmicro.com Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 01daf655e20b..b371066dd5bc 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -611,7 +611,7 @@ static int earpiece_event(struct snd_soc_dapm_widget *w, break; default: - BUG(); + WARN(1, "Invalid event %d\n", event); break; } -- cgit From a6ed0608bd289b45a9e42e3b0f33ff55bdac9b0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2013 11:07:19 +0100 Subject: ASoC: Replace BUG() with WARN() BUG() used in the driver is just to spit the stack trace on buggy points, not really needed to stop the whole operation. For that purpose, it'd be more convenient to use WARN() instead with more error information. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 6 ++++-- sound/soc/soc-dapm.c | 6 +++--- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index e72f55428f0b..223dc0636fe5 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -39,7 +39,8 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx, break; } default: - BUG(); + WARN(1, "Invalid word_size %d\n", word_size); + break; } return false; } @@ -60,7 +61,8 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, return cache[idx]; } default: - BUG(); + WARN(1, "Invalid word_size %d\n", word_size); + break; } /* unreachable */ return -1; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b2949aed1ac2..974a4ce14239 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1412,7 +1412,7 @@ static void dapm_seq_check_event(struct snd_soc_card *card, power = 0; break; default: - BUG(); + WARN(1, "Unknown event %d\n", event); return; } @@ -2001,7 +2001,7 @@ static ssize_t dapm_bias_read_file(struct file *file, char __user *user_buf, level = "Off\n"; break; default: - BUG(); + WARN(1, "Unknown bias_level %d\n", dapm->bias_level); level = "Unknown\n"; break; } @@ -3416,7 +3416,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, break; default: - BUG(); + WARN(1, "Unknown event %d\n", event); return -EINVAL; } -- cgit From 6c452bdac799e5ab94d658ea3517cfd57d832e6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2013 18:40:00 +0100 Subject: ASoC: wm_adsp: Fix BUG_ON() and WARN_ON() usages This patch does: - Move the sanity check with WARN_ON() in wm_adsp_region_to_reg() and remove the checks in the callers, - Fix wrong WARN_ON() usages, replaced with WARN(), - Fix unreachable or wrong BUG_ON() usages and replace with WARN_ON(). Signed-off-by: Takashi Iwai Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b38f3506418f..8dfce8f1ad2f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -341,6 +341,8 @@ static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, unsigned int offset) { + if (WARN_ON(!region)) + return offset; switch (region->type) { case WMFW_ADSP1_PM: return region->base + (offset * 3); @@ -353,7 +355,7 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, case WMFW_ADSP1_ZM: return region->base + (offset * 2); default: - WARN_ON(NULL != "Unknown memory region type"); + WARN(1, "Unknown memory region type"); return offset; } } @@ -602,7 +604,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) break; default: - BUG_ON(NULL == "Unknown DSP type"); + WARN(1, "Unknown DSP type"); goto out_fw; } @@ -642,27 +644,22 @@ static int wm_adsp_load(struct wm_adsp *dsp) reg = offset; break; case WMFW_ADSP1_PM: - BUG_ON(!mem); region_name = "PM"; reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_DM: - BUG_ON(!mem); region_name = "DM"; reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_XM: - BUG_ON(!mem); region_name = "XM"; reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_YM: - BUG_ON(!mem); region_name = "YM"; reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_ZM: - BUG_ON(!mem); region_name = "ZM"; reg = wm_adsp_region_to_reg(mem, offset); break; @@ -901,10 +898,8 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) break; } - if (mem == NULL) { - BUG_ON(mem != NULL); + if (WARN_ON(!mem)) return -EINVAL; - } switch (dsp->type) { case WMFW_ADSP1: @@ -998,7 +993,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) break; default: - BUG_ON(NULL == "Unknown DSP type"); + WARN(1, "Unknown DSP type"); return -EINVAL; } -- cgit From bf4edea863c435c302041cf8bb01c8b3ca729449 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2013 18:38:47 +0100 Subject: ASoC: dapm: Use WARN_ON() instead of BUG_ON() Leaving BUG_ON() in a core layer like dapm is rather inappropriate as it leads to panic(), even though sanity checks might be still useful for debugging. Instead, Use WARN_ON(), and handle the error cases accordingly. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 974a4ce14239..47dfe17ed4e8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1444,7 +1444,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, power_list)->reg; list_for_each_entry(w, pending, power_list) { - BUG_ON(reg != w->reg); + WARN_ON(reg != w->reg); w->power = w->new_power; mask |= w->mask << w->shift; @@ -3329,8 +3329,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, u64 fmt; int ret; - BUG_ON(!config); - BUG_ON(list_empty(&w->sources) || list_empty(&w->sinks)); + if (WARN_ON(!config) || + WARN_ON(list_empty(&w->sources) || list_empty(&w->sinks))) + return -EINVAL; /* We only support a single source and sink, pick the first */ source_p = list_first_entry(&w->sources, struct snd_soc_dapm_path, @@ -3338,9 +3339,10 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, list_source); - BUG_ON(!source_p || !sink_p); - BUG_ON(!sink_p->source || !source_p->sink); - BUG_ON(!source_p->source || !sink_p->sink); + if (WARN_ON(!source_p || !sink_p) || + WARN_ON(!sink_p->source || !source_p->sink) || + WARN_ON(!source_p->source || !sink_p->sink)) + return -EINVAL; source = source_p->source->priv; sink = sink_p->sink->priv; -- cgit From ca2b029515b4a81bbedcffc69a5d9574bb5ba475 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 7 Nov 2013 14:45:16 +0800 Subject: ASoC: generic-dmaengine-pcm: Use SNDRV_DMA_TYPE_DEV_IRAM as default When allocating memory space for DMA buffer, use on-chip internal SRAM as default choice to save power. Since the core would allocate memory from traditional external memory if iram allocation failed, we don't need to worry about any side effect. Signed-off-by: Nicolin Chen Acked-by: Lars-Peter Clausen Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ee0790337ec9..51fded3640a5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -228,7 +228,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) } ret = snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, + SNDRV_DMA_TYPE_DEV_IRAM, dmaengine_dma_dev(pcm, substream), prealloc_buffer_size, max_buffer_size); -- cgit From 3b70a67da0bebca02a130eff8084eee457a2b836 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Thu, 7 Nov 2013 11:55:15 -0800 Subject: ALSA: hda_intel: ratelimit "spurious response" message dmesg here has a 100+ consecutive lines of: [ 1464.219446] hda-intel 0000:00:14.2: spurious response 0x0:0x0, last cmd=0x170500 [ 1464.219451] hda-intel 0000:00:14.2: spurious response 0x0:0x0, last cmd=0x170500 [ 1464.219454] hda-intel 0000:00:14.2: spurious response 0x0:0x0, last cmd=0x170500 ... Ratelimit the message to reduce the dmesg log noise. Coalesce the format while at it. Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 17f4aa8cddd5..7a09404579a7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -914,12 +914,12 @@ static void azx_update_rirb(struct azx *chip) chip->rirb.res[addr] = res; smp_wmb(); chip->rirb.cmds[addr]--; - } else - snd_printk(KERN_ERR SFX "%s: spurious response %#x:%#x, " - "last cmd=%#08x\n", + } else if (printk_ratelimit()) { + snd_printk(KERN_ERR SFX "%s: spurious response %#x:%#x, last cmd=%#08x\n", pci_name(chip->pci), res, res_ex, chip->last_cmd[addr]); + } } } -- cgit From 380702192f616df5b2ba704b85cc479a3da92c91 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 1 Nov 2013 15:57:35 +0800 Subject: ALSA: hda/realtek - Make fixup regs persist after resume Upon suspend / resume, the fixup register settings are lost because sending HDA_FIXUP_ACT_PRE_PROBE is not part of the resume path. Instead, write our registers in response to the HDA_FIXUP_ACT_INIT, which happens after initial probe and upon resume. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index daf720502385..6a214662700e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3503,6 +3503,8 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, /* Disable AA-loopback as it causes white noise */ spec->gen.mixer_nid = 0; spec->gen.hp_automute_hook = alc283_hp_automute_hook; + break; + case HDA_FIXUP_ACT_INIT: /* MIC2-VREF control */ /* Set to manual mode */ val = alc_read_coef_idx(codec, 0x06); -- cgit From 10227a94874435b038d5303ca05c1e158927084b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 8 Nov 2013 00:55:00 -0200 Subject: ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable Commit 68f9672b (ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation) introduced the following build warning: sound/soc/fsl/imx-pcm-fiq.c:53:26: warning: unused variable 'runtime' [-Wunused-variable] Remove the unused 'runtime' variable. Signed-off-by: Fabio Estevam Acked-by: Oskar Schirmer Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-fiq.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 94720a6dd7b6..10e330514ed8 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -50,7 +50,6 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct imx_pcm_runtime_data *iprtd = container_of(hrt, struct imx_pcm_runtime_data, hrt); struct snd_pcm_substream *substream = iprtd->substream; - struct snd_pcm_runtime *runtime = substream->runtime; struct pt_regs regs; if (!atomic_read(&iprtd->running)) -- cgit From ea8e5e591801b0d0c25dc0a0f1f5b508be7004cb Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 8 Nov 2013 12:54:44 +0300 Subject: ASoC: fsl: imx-wm8962: remove an unneeded check "data->codec_clk" can't be an ERR_PTR here so I have removed the superflous check. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/fsl/imx-wm8962.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 8e5b2c6a16d9..f7be44ad443a 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -279,8 +279,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) return 0; clk_fail: - if (!IS_ERR(data->codec_clk)) - clk_disable_unprepare(data->codec_clk); + clk_disable_unprepare(data->codec_clk); fail: if (ssi_np) of_node_put(ssi_np); -- cgit From 885845d78551be7bf8570f6283df8b7a7797c4d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2013 12:50:31 +0100 Subject: ALSA: hda - Apply MacBook fixups for CS4208 correctly The commit [8fe7b65ab465: ALSA: hda - Apply GPIO setup for MacBooks with CS4208] added a fixup entry matching with the vendor id 0x106b. This broke the fixups for previous MBA6,1 and 6,2, since the PCI SSID vendor id matches before evaluating the codec SSIDs. We had a similar issue on Mac with Sigmatel codecs, and solve this problem again similarly, by introducing a skeleton entry matching with the all MacBooks, then remap to the right one. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64401 Fixes: 8fe7b65ab465 ('ALSA: hda - Apply GPIO setup for MacBooks with CS4208') Cc: [v3.12+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 27 +++++++++++++++++++++++++-- 1 file changed, 25 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1ce1f4067b15..072755c8289c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -597,6 +597,7 @@ static int patch_cs420x(struct hda_codec *codec) * Its layout is no longer compatible with CS4206/CS4207 */ enum { + CS4208_MAC_AUTO, CS4208_MBA6, CS4208_GPIO0, }; @@ -608,10 +609,14 @@ static const struct hda_model_fixup cs4208_models[] = { }; static const struct snd_pci_quirk cs4208_fixup_tbl[] = { - /* codec SSID */ + SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS4208_MAC_AUTO), + {} /* terminator */ +}; + +/* codec SSID matching */ +static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6), SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6), - SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS4208_GPIO0), {} /* terminator */ }; @@ -627,6 +632,20 @@ static void cs4208_fixup_gpio0(struct hda_codec *codec, } } +static const struct hda_fixup cs4208_fixups[]; + +/* remap the fixup from codec SSID and apply it */ +static void cs4208_fixup_mac(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + snd_hda_pick_fixup(codec, NULL, cs4208_mac_fixup_tbl, cs4208_fixups); + if (codec->fixup_id < 0 || codec->fixup_id == CS4208_MAC_AUTO) + codec->fixup_id = CS4208_GPIO0; /* default fixup */ + snd_hda_apply_fixup(codec, action); +} + static const struct hda_fixup cs4208_fixups[] = { [CS4208_MBA6] = { .type = HDA_FIXUP_PINS, @@ -638,6 +657,10 @@ static const struct hda_fixup cs4208_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cs4208_fixup_gpio0, }, + [CS4208_MAC_AUTO] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs4208_fixup_mac, + }, }; /* correct the 0dB offset of input pins */ -- cgit From 9a22a8f558d09a83965d2bbe168294eb8ffb70e9 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 8 Nov 2013 15:54:49 +0800 Subject: ALSA: hda/realtek - Add new codec ALC255/ALC3234 UAJ supported New codec ALC255/ALC3234 support multifunction jacks. It used for menual select the input device. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 104 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 97 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a214662700e..24d924d563aa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -577,26 +577,35 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) /* * COEF access helper functions */ -static int alc_read_coef_idx(struct hda_codec *codec, - unsigned int coef_idx) + +static int alc_read_coefex_idx(struct hda_codec *codec, + hda_nid_t nid, + unsigned int coef_idx) { unsigned int val; - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, coef_idx); - val = snd_hda_codec_read(codec, 0x20, 0, + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0); return val; } -static void alc_write_coef_idx(struct hda_codec *codec, unsigned int coef_idx, +#define alc_read_coef_idx(codec, coef_idx) \ + alc_read_coefex_idx(codec, 0x20, coef_idx) + +static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_val) { - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, coef_idx); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PROC_COEF, coef_val); } +#define alc_write_coef_idx(codec, coef_idx, coef_val) \ + alc_write_coefex_idx(codec, 0x20, coef_idx, coef_val) + /* a special bypass for COEF 0; read the cached value at the second time */ static unsigned int alc_get_coef0(struct hda_codec *codec) { @@ -3109,6 +3118,19 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) int val; switch (codec->vendor_id) { + case 0x10ec0255: + /* LDO and MISC control */ + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + /* UAJ function set to menual mode */ + alc_write_coef_idx(codec, 0x45, 0xd089); + /* Direct Drive HP Amp control(Set to verb control)*/ + val = alc_read_coefex_idx(codec, 0x57, 0x05); + alc_write_coefex_idx(codec, 0x57, 0x05, val & ~(1<<14)); + /* Set MIC2 Vref gate with HP */ + alc_write_coef_idx(codec, 0x06, 0x6104); + /* Direct Drive HP Amp control */ + alc_write_coefex_idx(codec, 0x57, 0x03, 0x8aa6); + break; case 0x10ec0283: alc_write_coef_idx(codec, 0x1b, 0x0c0b); alc_write_coef_idx(codec, 0x45, 0xc429); @@ -3140,6 +3162,14 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, int val; switch (codec->vendor_id) { + case 0x10ec0255: + alc_write_coef_idx(codec, 0x45, 0xc489); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coefex_idx(codec, 0x57, 0x03, 0x8aa6); + /* Set MIC2 Vref gate to normal */ + alc_write_coef_idx(codec, 0x06, 0x6100); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; case 0x10ec0283: alc_write_coef_idx(codec, 0x45, 0xc429); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); @@ -3171,6 +3201,12 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, static void alc_headset_mode_default(struct hda_codec *codec) { switch (codec->vendor_id) { + case 0x10ec0255: + alc_write_coef_idx(codec, 0x45, 0xc089); + alc_write_coef_idx(codec, 0x45, 0xc489); + alc_write_coefex_idx(codec, 0x57, 0x03, 0x8ea6); + alc_write_coef_idx(codec, 0x49, 0x0049); + break; case 0x10ec0283: alc_write_coef_idx(codec, 0x06, 0x2100); alc_write_coef_idx(codec, 0x32, 0x4ea3); @@ -3194,6 +3230,12 @@ static void alc_headset_mode_default(struct hda_codec *codec) static void alc_headset_mode_ctia(struct hda_codec *codec) { switch (codec->vendor_id) { + case 0x10ec0255: + /* Set to CTIA type */ + alc_write_coef_idx(codec, 0x45, 0xd489); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coefex_idx(codec, 0x57, 0x03, 0x8ea6); + break; case 0x10ec0283: alc_write_coef_idx(codec, 0x45, 0xd429); alc_write_coef_idx(codec, 0x1b, 0x0c2b); @@ -3216,6 +3258,12 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) static void alc_headset_mode_omtp(struct hda_codec *codec) { switch (codec->vendor_id) { + case 0x10ec0255: + /* Set to OMTP Type */ + alc_write_coef_idx(codec, 0x45, 0xe489); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coefex_idx(codec, 0x57, 0x03, 0x8ea6); + break; case 0x10ec0283: alc_write_coef_idx(codec, 0x45, 0xe429); alc_write_coef_idx(codec, 0x1b, 0x0c2b); @@ -3241,6 +3289,15 @@ static void alc_determine_headset_type(struct hda_codec *codec) struct alc_spec *spec = codec->spec; switch (codec->vendor_id) { + case 0x10ec0255: + /* combo jack auto switch control(Check type)*/ + alc_write_coef_idx(codec, 0x45, 0xd089); + /* combo jack auto switch control(Vref conteol) */ + alc_write_coef_idx(codec, 0x49, 0x0149); + msleep(300); + val = alc_read_coef_idx(codec, 0x46); + is_ctia = (val & 0x0070) == 0x0070; + break; case 0x10ec0283: alc_write_coef_idx(codec, 0x45, 0xd029); msleep(300); @@ -3387,6 +3444,21 @@ static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + /* Set to iphone type */ + alc_write_coef_idx(codec, 0x1b, 0x880b); + alc_write_coef_idx(codec, 0x45, 0xd089); + alc_write_coef_idx(codec, 0x1b, 0x080b); + alc_write_coef_idx(codec, 0x46, 0x0004); + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + msleep(30); + } + alc_fixup_headset_mode(codec, fix, action); +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3688,6 +3760,8 @@ enum { ALC283_FIXUP_INT_MIC, ALC290_FIXUP_MONO_SPEAKERS, ALC269_FIXUP_THINKPAD_ACPI, + ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_FIXUP_HEADSET_MODE, }; static const struct hda_fixup alc269_fixups[] = { @@ -3997,6 +4071,20 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_HEADSET_MODE + }, + [ALC255_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc255, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4039,6 +4127,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit