diff options
Diffstat (limited to 'sound')
81 files changed, 1737 insertions, 706 deletions
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 4e7ec2b49873..d0f00356fc11 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -101,7 +101,7 @@ void *snd_malloc_sgbuf_pages(struct device *device, if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device, chunk, &tmpb) < 0) { if (!sgbuf->pages) - return NULL; + goto _failed; if (!res_size) goto _failed; size = sgbuf->pages * PAGE_SIZE; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4f502a2bdc3c..0a436626182b 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst for (page = blk->first_page; page <= blk->last_page; page++, idx++) { unsigned long ofs = idx << PAGE_SHIFT; dma_addr_t addr; - addr = snd_pcm_sgbuf_get_addr(substream, ofs); + if (ofs >= runtime->dma_bytes) + addr = emu->silent_page.addr; + else + addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { printk(KERN_ERR "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 647218d69f68..4f7d2dfcef7b 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, if (cfg->dig_outs) snd_printd(" dig-out=0x%x/0x%x\n", cfg->dig_out_pins[0], cfg->dig_out_pins[1]); - snd_printd(" inputs:"); + snd_printd(" inputs:\n"); for (i = 0; i < cfg->num_inputs; i++) { - snd_printd(" %s=0x%x", + snd_printd(" %s=0x%x\n", hda_get_autocfg_input_label(codec, cfg, i), cfg->inputs[i].pin); } - snd_printd("\n"); if (cfg->dig_in_pin) snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 14361184ae1e..5e22a8f43d2e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2967,12 +2967,10 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO), SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), @@ -2988,14 +2986,10 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), - SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), - SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 69b928449789..8f23374fa642 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -877,8 +877,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; - hinfo->nid = 0; /* clear the leftover value */ - /* Validate hinfo */ pin_idx = hinfo_to_pin_index(spec, hinfo); if (snd_BUG_ON(pin_idx < 0)) @@ -1163,6 +1161,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } +static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_cleanup_stream(codec, hinfo->nid); + return 0; +} + static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1202,6 +1208,7 @@ static const struct hda_pcm_ops generic_ops = { .open = hdmi_pcm_open, .close = hdmi_pcm_close, .prepare = generic_hdmi_playback_pcm_prepare, + .cleanup = generic_hdmi_playback_pcm_cleanup, }; static int generic_hdmi_build_pcms(struct hda_codec *codec) @@ -1220,7 +1227,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; pstr->substreams = 1; pstr->ops = generic_ops; - pstr->nid = 1; /* FIXME: just for avoiding a debug WARNING */ /* other pstr fields are set in open */ } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 344b221d2102..4f81dd44c837 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6099,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_PCM_44K] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_pcm_44k, + .chained = true, + .chain_id = ALC269_FIXUP_QUANTA_MUTE }, [ALC269_FIXUP_STEREO_DMIC] = { .type = ALC_FIXUP_FUNC, @@ -6206,9 +6208,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), #if 0 diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index c5de0a84566f..c24de902f5f5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,6 +9,7 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER + select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 00a555a743b6..c1264007b4ee 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-io.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d542d4063771..16b9c9efd19a 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -59,62 +59,63 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { #define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) -static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.0", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.1", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .codec_dai_name = "ad1836-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }; static struct snd_soc_card bf5xx_ad1836 = { .name = "bfin-ad1836", .owner = THIS_MODULE, - .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM], + .dai_link = &bf5xx_ad1836_dai, .num_links = 1, }; -static struct platform_device *bfxx_ad1836_snd_device; - -static int __init bf5xx_ad1836_init(void) +static __devinit int bf5xx_ad1836_driver_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &bf5xx_ad1836; + const char **link_name; int ret; - bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1836_snd_device) - return -ENOMEM; + link_name = pdev->dev.platform_data; + if (!link_name) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + bf5xx_ad1836_dai.cpu_dai_name = link_name[0]; + bf5xx_ad1836_dai.codec_name = link_name[1]; - platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836); - ret = platform_device_add(bfxx_ad1836_snd_device); + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + ret = snd_soc_register_card(card); if (ret) - platform_device_put(bfxx_ad1836_snd_device); - + dev_err(&pdev->dev, "Failed to register card\n"); return ret; } -static void __exit bf5xx_ad1836_exit(void) +static int __devexit bf5xx_ad1836_driver_remove(struct platform_device *pdev) { - platform_device_unregister(bfxx_ad1836_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(bf5xx_ad1836_init); -module_exit(bf5xx_ad1836_exit); +static struct platform_driver bf5xx_ad1836_driver = { + .driver = { + .name = "bfin-snd-ad1836", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bf5xx_ad1836_driver_probe, + .remove = __devexit_p(bf5xx_ad1836_driver_remove), +}; +module_platform_driver(bf5xx_ad1836_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3c795921c5f6..b7836503dc69 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2404,8 +2404,12 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); + /* Inform SoC Core that we have our own I/O arrangements. */ + codec->control_data = (void *)true; + /* Setup AB8500 according to board-settings */ - pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + pdata = dev_get_platdata(dev->parent); + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 8c39dddd7d00..11b1b714b8b5 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 44f59064d8de..704544bfc90d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1392,17 +1392,7 @@ static struct i2c_driver adau1373_i2c_driver = { .id_table = adau1373_i2c_id, }; -static int __init adau1373_init(void) -{ - return i2c_add_driver(&adau1373_i2c_driver); -} -module_init(adau1373_init); - -static void __exit adau1373_exit(void) -{ - i2c_del_driver(&adau1373_i2c_driver); -} -module_exit(adau1373_exit); +module_i2c_driver(adau1373_i2c_driver); MODULE_DESCRIPTION("ASoC ADAU1373 driver"); MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 3d50fc8646b6..51f2f3cd8136 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -527,17 +527,7 @@ static struct i2c_driver adau1701_i2c_driver = { .id_table = adau1701_i2c_id, }; -static int __init adau1701_init(void) -{ - return i2c_add_driver(&adau1701_i2c_driver); -} -module_init(adau1701_init); - -static void __exit adau1701_exit(void) -{ - i2c_del_driver(&adau1701_i2c_driver); -} -module_exit(adau1701_exit); +module_i2c_driver(adau1701_i2c_driver); MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver"); MODULE_AUTHOR("Cliff Cai <cliff.cai@analog.com>"); diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 5fb7c2a80e6d..2b457976a7bf 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -696,17 +696,7 @@ static struct i2c_driver ak4671_i2c_driver = { .id_table = ak4671_i2c_id, }; -static int __init ak4671_modinit(void) -{ - return i2c_add_driver(&ak4671_i2c_driver); -} -module_init(ak4671_modinit); - -static void __exit ak4671_exit(void) -{ - i2c_del_driver(&ak4671_i2c_driver); -} -module_exit(ak4671_exit); +module_i2c_driver(ak4671_i2c_driver); MODULE_DESCRIPTION("ASoC AK4671 codec driver"); MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>"); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9cacaf2d52..5e96a0a1669c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -229,6 +229,69 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_out_ev); +static unsigned int arizona_sysclk_48k_rates[] = { + 6144000, + 12288000, + 22579200, + 49152000, +}; + +static unsigned int arizona_sysclk_44k1_rates[] = { + 5644800, + 11289600, + 24576000, + 45158400, +}; + +static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, + unsigned int freq) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + unsigned int *rates; + int ref, div, refclk; + + switch (clk) { + case ARIZONA_CLK_OPCLK: + reg = ARIZONA_OUTPUT_SYSTEM_CLOCK; + refclk = priv->sysclk; + break; + case ARIZONA_CLK_ASYNC_OPCLK: + reg = ARIZONA_OUTPUT_ASYNC_CLOCK; + refclk = priv->asyncclk; + break; + default: + return -EINVAL; + } + + if (refclk % 8000) + rates = arizona_sysclk_44k1_rates; + else + rates = arizona_sysclk_48k_rates; + + for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) && + rates[ref] <= refclk; ref++) { + div = 1; + while (rates[ref] / div >= freq && div < 32) { + if (rates[ref] / div == freq) { + dev_dbg(codec->dev, "Configured %dHz OPCLK\n", + freq); + snd_soc_update_bits(codec, reg, + ARIZONA_OPCLK_DIV_MASK | + ARIZONA_OPCLK_SEL_MASK, + (div << + ARIZONA_OPCLK_DIV_SHIFT) | + ref); + return 0; + } + div++; + } + } + + dev_err(codec->dev, "Unable to generate %dHz OPCLK\n", freq); + return -EINVAL; +} + int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { @@ -252,6 +315,9 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, reg = ARIZONA_ASYNC_CLOCK_1; clk = &priv->asyncclk; break; + case ARIZONA_CLK_OPCLK: + case ARIZONA_CLK_ASYNC_OPCLK: + return arizona_set_opclk(codec, clk_id, freq); default: return -EINVAL; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 59caca8865e8..eb66b52777c9 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -17,8 +17,10 @@ #include <sound/soc.h> -#define ARIZONA_CLK_SYSCLK 1 -#define ARIZONA_CLK_ASYNCCLK 2 +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 +#define ARIZONA_CLK_OPCLK 3 +#define ARIZONA_CLK_ASYNC_OPCLK 4 #define ARIZONA_CLK_SRC_MCLK1 0x0 #define ARIZONA_CLK_SRC_MCLK2 0x1 diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 047917f0b8ae..44a176f74170 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -29,6 +29,8 @@ #include <linux/i2c.h> #include <linux/delay.h> #include <linux/regulator/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> /* * The codec isn't really big-endian or little-endian, since the I2S @@ -639,6 +641,15 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .reg_cache_default = cs4270_default_reg_cache, }; +/* + * cs4270_of_match - the device tree bindings + */ +static const struct of_device_id cs4270_of_match[] = { + { .compatible = "cirrus,cs4270", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4270_of_match); + /** * cs4270_i2c_probe - initialize the I2C interface of the CS4270 * @i2c_client: the I2C client object @@ -650,9 +661,25 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { + struct device_node *np = i2c_client->dev.of_node; struct cs4270_private *cs4270; int ret; + /* See if we have a way to bring the codec out of reset */ + if (np) { + enum of_gpio_flags flags; + int gpio = of_get_named_gpio_flags(np, "reset-gpio", 0, &flags); + + if (gpio_is_valid(gpio)) { + ret = devm_gpio_request_one(&i2c_client->dev, gpio, + flags & OF_GPIO_ACTIVE_LOW ? + GPIOF_OUT_INIT_LOW : GPIOF_OUT_INIT_HIGH, + "cs4270 reset"); + if (ret < 0) + return ret; + } + } + /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); @@ -718,23 +745,14 @@ static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", .owner = THIS_MODULE, + .of_match_table = cs4270_of_match, }, .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, }; -static int __init cs4270_init(void) -{ - return i2c_add_driver(&cs4270_i2c_driver); -} -module_init(cs4270_init); - -static void __exit cs4270_exit(void) -{ - i2c_del_driver(&cs4270_i2c_driver); -} -module_exit(cs4270_exit); +module_i2c_driver(cs4270_i2c_driver); MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 091d0193f507..1e0fa3b5f79a 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -614,24 +614,7 @@ static struct i2c_driver cs42l51_i2c_driver = { .remove = cs42l51_i2c_remove, }; -static int __init cs42l51_init(void) -{ - int ret; - - ret = i2c_add_driver(&cs42l51_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver\n", __func__); - return ret; - } - return 0; -} -module_init(cs42l51_init); - -static void __exit cs42l51_exit(void) -{ - i2c_del_driver(&cs42l51_i2c_driver); -} -module_exit(cs42l51_exit); +module_i2c_driver(cs42l51_i2c_driver); MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 628daf6a1d97..61599298fb26 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -24,7 +24,6 @@ #include <linux/slab.h> #include <linux/workqueue.h> #include <linux/platform_device.h> -#include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5d8f39e32978..1bf55602c9eb 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -13,7 +13,6 @@ */ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index ba4fafb93e56..81a328c78838 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -250,17 +250,7 @@ static struct i2c_driver lm4857_i2c_driver = { .id_table = lm4857_i2c_id, }; -static int __init lm4857_init(void) -{ - return i2c_add_driver(&lm4857_i2c_driver); -} -module_init(lm4857_init); - -static void __exit lm4857_exit(void) -{ - i2c_del_driver(&lm4857_i2c_driver); -} -module_exit(lm4857_exit); +module_i2c_driver(lm4857_i2c_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); MODULE_DESCRIPTION("LM4857 amplifier driver"); diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index af7324b79dd0..3264a5169306 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2107,23 +2107,7 @@ static struct i2c_driver max98088_i2c_driver = { .id_table = max98088_i2c_id, }; -static int __init max98088_init(void) -{ - int ret; - - ret = i2c_add_driver(&max98088_i2c_driver); - if (ret) - pr_err("Failed to register max98088 I2C driver: %d\n", ret); - - return ret; -} -module_init(max98088_init); - -static void __exit max98088_exit(void) -{ - i2c_del_driver(&max98088_i2c_driver); -} -module_exit(max98088_exit); +module_i2c_driver(max98088_i2c_driver); MODULE_DESCRIPTION("ALSA SoC MAX98088 driver"); MODULE_AUTHOR("Peter Hsiang, Jesse Marroquin"); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 7cd508e16a5c..38d43c59d3f4 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2533,23 +2533,7 @@ static struct i2c_driver max98095_i2c_driver = { .id_table = max98095_i2c_id, }; -static int __init max98095_init(void) -{ - int ret; - - ret = i2c_add_driver(&max98095_i2c_driver); - if (ret) - pr_err("Failed to register max98095 I2C driver: %d\n", ret); - - return ret; -} -module_init(max98095_init); - -static void __exit max98095_exit(void) -{ - i2c_del_driver(&max98095_i2c_driver); -} -module_exit(max98095_exit); +module_i2c_driver(max98095_i2c_driver); MODULE_DESCRIPTION("ALSA SoC MAX98095 driver"); MODULE_AUTHOR("Peter Hsiang"); diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index a1913091f56c..efe535c37b39 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -369,17 +369,7 @@ static struct i2c_driver max9850_i2c_driver = { .id_table = max9850_i2c_id, }; -static int __init max9850_init(void) -{ - return i2c_add_driver(&max9850_i2c_driver); -} -module_init(max9850_init); - -static void __exit max9850_exit(void) -{ - i2c_del_driver(&max9850_i2c_driver); -} -module_exit(max9850_exit); +module_i2c_driver(max9850_i2c_driver); MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp@taskit.de>"); MODULE_DESCRIPTION("ASoC MAX9850 codec driver"); diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 3a2ba3d8fd6d..d15e5943c85e 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -291,17 +291,7 @@ static struct i2c_driver max9877_i2c_driver = { .id_table = max9877_i2c_id, }; -static int __init max9877_init(void) -{ - return i2c_add_driver(&max9877_i2c_driver); -} -module_init(max9877_init); - -static void __exit max9877_exit(void) -{ - i2c_del_driver(&max9877_i2c_driver); -} -module_exit(max9877_exit); +module_i2c_driver(max9877_i2c_driver); MODULE_DESCRIPTION("ASoC MAX9877 amp driver"); MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>"); diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 6276e352125f..8f726c063f42 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->mc13xxx; + mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8af6a5245b18..df2f99d1d428 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ + {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ @@ -1357,8 +1358,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_dapm_new_widgets(&codec->dapm); - return 0; err: diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 8d717f4b5a87..51b7313a4c14 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1006,17 +1006,7 @@ static struct i2c_driver sta32x_i2c_driver = { .id_table = sta32x_i2c_id, }; -static int __init sta32x_init(void) -{ - return i2c_add_driver(&sta32x_i2c_driver); -} -module_init(sta32x_init); - -static void __exit sta32x_exit(void) -{ - i2c_del_driver(&sta32x_i2c_driver); -} -module_exit(sta32x_exit); +module_i2c_driver(sta32x_i2c_driver); MODULE_DESCRIPTION("ASoC STA32X driver"); MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 0c225cd569d2..9e3144862386 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -358,7 +358,7 @@ static int sta529_resume(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sta529_codec_driver = { +static const struct snd_soc_codec_driver sta529_codec_driver = { .probe = sta529_probe, .remove = sta529_remove, .set_bias_level = sta529_set_bias_level, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 982e437799a8..33c0f3d39c87 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index b0a73d37ed52..f230292ba96b 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -746,24 +746,7 @@ static struct i2c_driver aic32x4_i2c_driver = { .id_table = aic32x4_i2c_id, }; -static int __init aic32x4_modinit(void) -{ - int ret = 0; - - ret = i2c_add_driver(&aic32x4_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(aic32x4_modinit); - -static void __exit aic32x4_exit(void) -{ - i2c_del_driver(&aic32x4_i2c_driver); -} -module_exit(aic32x4_exit); +module_i2c_driver(aic32x4_i2c_driver); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>"); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index dc78f5a4bcbf..01485bd51404 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1499,23 +1499,7 @@ static struct i2c_driver aic3x_i2c_driver = { .id_table = aic3x_i2c_id, }; -static int __init aic3x_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&aic3x_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC3x I2C driver: %d\n", - ret); - } - return ret; -} -module_init(aic3x_modinit); - -static void __exit aic3x_exit(void) -{ - i2c_del_driver(&aic3x_i2c_driver); -} -module_exit(aic3x_exit); +module_i2c_driver(aic3x_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver"); MODULE_AUTHOR("Vladimir Barinov"); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 0dd41077ab79..d2e16c5d7d1f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1621,24 +1621,7 @@ static struct i2c_driver tlv320dac33_i2c_driver = { .id_table = tlv320dac33_i2c_id, }; -static int __init dac33_module_init(void) -{ - int r; - r = i2c_add_driver(&tlv320dac33_i2c_driver); - if (r < 0) { - printk(KERN_ERR "DAC33: driver registration failed\n"); - return r; - } - return 0; -} -module_init(dac33_module_init); - -static void __exit dac33_module_exit(void) -{ - i2c_del_driver(&tlv320dac33_i2c_driver); -} -module_exit(dac33_module_exit); - +module_i2c_driver(tlv320dac33_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>"); diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 6fe4aa3ac544..565ff39ad3a3 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -487,19 +487,8 @@ static struct i2c_driver tpa6130a2_i2c_driver = { .id_table = tpa6130a2_id, }; -static int __init tpa6130a2_init(void) -{ - return i2c_add_driver(&tpa6130a2_i2c_driver); -} - -static void __exit tpa6130a2_exit(void) -{ - i2c_del_driver(&tpa6130a2_i2c_driver); -} +module_i2c_driver(tpa6130a2_i2c_driver); MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>"); MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); MODULE_LICENSE("GPL"); - -module_init(tpa6130a2_init); -module_exit(tpa6130a2_exit); diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 3fd5b29dc933..89cd6fcad015 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -858,17 +858,7 @@ static struct i2c_driver wm2000_i2c_driver = { .id_table = wm2000_i2c_id, }; -static int __init wm2000_init(void) -{ - return i2c_add_driver(&wm2000_i2c_driver); -} -module_init(wm2000_init); - -static void __exit wm2000_exit(void) -{ - i2c_del_driver(&wm2000_i2c_driver); -} -module_exit(wm2000_exit); +module_i2c_driver(wm2000_i2c_driver); MODULE_DESCRIPTION("ASoC WM2000 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>"); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 32682c1b7cde..71debd0a3822 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2270,17 +2270,7 @@ static struct i2c_driver wm2200_i2c_driver = { .id_table = wm2200_i2c_id, }; -static int __init wm2200_modinit(void) -{ - return i2c_add_driver(&wm2200_i2c_driver); -} -module_init(wm2200_modinit); - -static void __exit wm2200_exit(void) -{ - i2c_del_driver(&wm2200_i2c_driver); -} -module_exit(wm2200_exit); +module_i2c_driver(wm2200_i2c_driver); MODULE_DESCRIPTION("ASoC WM2200 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16d383e..2e6f1ffc9fd4 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -280,11 +280,36 @@ ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const char *wm5102_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "EPOUT", + "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm5102_aec_loopback_values[] = { + 0, 1, 2, 3, 4, 6, 7, 8, 9, +}; + +static const struct soc_enum wm5102_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, + ARIZONA_AEC_LOOPBACK_SRC_MASK, + ARRAY_SIZE(wm5102_aec_loopback_texts), + wm5102_aec_loopback_texts, + wm5102_aec_loopback_values); + +static const struct snd_kcontrol_new wm5102_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm5102_aec_loopback); + static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), @@ -431,6 +456,9 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), @@ -528,6 +556,7 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "Noise Generator", "Noise Generator" }, \ { name, "Tone Generator 1", "Tone Generator 1" }, \ { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "AEC", "AEC Loopback" }, \ { name, "IN1L", "IN1L PGA" }, \ { name, "IN1R", "IN1R PGA" }, \ { name, "IN2L", "IN2L PGA" }, \ @@ -688,21 +717,30 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MIXER_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", "EPOUT", "OUT3L" }, { "EPOUTN", NULL, "OUT3L" }, { "EPOUTP", 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" }, }; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f7065189..3db6e6e7a591 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -305,6 +305,10 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 73f1c8d7bafb..839414f9e2ed 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2241,23 +2241,7 @@ static struct i2c_driver wm8903_i2c_driver = { .id_table = wm8903_i2c_id, }; -static int __init wm8903_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8903_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8903 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8903_modinit); - -static void __exit wm8903_exit(void) -{ - i2c_del_driver(&wm8903_i2c_driver); -} -module_exit(wm8903_exit); +module_i2c_driver(wm8903_i2c_driver); MODULE_DESCRIPTION("ASoC WM8903 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>"); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 481a3d9cfe48..b20aa4e7c3f9 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -785,23 +785,7 @@ static struct i2c_driver wm8940_i2c_driver = { .id_table = wm8940_i2c_id, }; -static int __init wm8940_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8940_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8940 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8940_modinit); - -static void __exit wm8940_exit(void) -{ - i2c_del_driver(&wm8940_i2c_driver); -} -module_exit(wm8940_exit); +module_i2c_driver(wm8940_i2c_driver); MODULE_DESCRIPTION("ASoC WM8940 driver"); MODULE_AUTHOR("Jonathan Cameron"); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 61fe97433e73..2f1c075755b1 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1071,23 +1071,7 @@ static struct i2c_driver wm8955_i2c_driver = { .id_table = wm8955_i2c_id, }; -static int __init wm8955_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8955_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8955 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8955_modinit); - -static void __exit wm8955_exit(void) -{ - i2c_del_driver(&wm8955_i2c_driver); -} -module_exit(wm8955_exit); +module_i2c_driver(wm8955_i2c_driver); MODULE_DESCRIPTION("ASoC WM8955 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 1332692ef81b..00121ba36597 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -946,7 +946,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->mbc_texts = kmalloc(sizeof(char *) * pdata->num_mbc_cfgs, GFP_KERNEL); if (!wm8994->mbc_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d MBC config texts\n", pdata->num_mbc_cfgs); return; @@ -958,9 +958,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->mbc_enum.max = pdata->num_mbc_cfgs; wm8994->mbc_enum.texts = wm8994->mbc_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add MBC mode controls: %d\n", ret); } @@ -974,7 +975,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_texts = kmalloc(sizeof(char *) * pdata->num_vss_cfgs, GFP_KERNEL); if (!wm8994->vss_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d VSS config texts\n", pdata->num_vss_cfgs); return; @@ -986,9 +987,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_enum.max = pdata->num_vss_cfgs; wm8994->vss_enum.texts = wm8994->vss_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add VSS mode controls: %d\n", ret); } @@ -1003,7 +1005,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_hpf_texts = kmalloc(sizeof(char *) * pdata->num_vss_hpf_cfgs, GFP_KERNEL); if (!wm8994->vss_hpf_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d VSS HPF config texts\n", pdata->num_vss_hpf_cfgs); return; @@ -1015,9 +1017,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs; wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add VSS HPFmode controls: %d\n", ret); } @@ -1033,7 +1036,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->enh_eq_texts = kmalloc(sizeof(char *) * pdata->num_enh_eq_cfgs, GFP_KERNEL); if (!wm8994->enh_eq_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d enhanced EQ config texts\n", pdata->num_enh_eq_cfgs); return; @@ -1045,9 +1048,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs; wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add enhanced EQ controls: %d\n", ret); } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 96518ac8e24c..804f4116912f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1010,23 +1010,7 @@ static struct i2c_driver wm8960_i2c_driver = { .id_table = wm8960_i2c_id, }; -static int __init wm8960_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8960_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8960_modinit); - -static void __exit wm8960_exit(void) -{ - i2c_del_driver(&wm8960_i2c_driver); -} -module_exit(wm8960_exit); +module_i2c_driver(wm8960_i2c_driver); MODULE_DESCRIPTION("ASoC WM8960 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 01edbcc754d2..719fb69a17c8 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1114,23 +1114,7 @@ static struct i2c_driver wm8961_i2c_driver = { .id_table = wm8961_i2c_id, }; -static int __init wm8961_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8961_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8961 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8961_modinit); - -static void __exit wm8961_exit(void) -{ - i2c_del_driver(&wm8961_i2c_driver); -} -module_exit(wm8961_exit); +module_i2c_driver(wm8961_i2c_driver); MODULE_DESCRIPTION("ASoC WM8961 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index eaf65863ec21..aa9ce9dd7d8a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*250k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x100); + + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + msleep(100); break; case SND_SOC_BIAS_OFF: diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index eef783f6b6d6..5ce647758443 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -721,23 +721,7 @@ static struct i2c_driver wm8971_i2c_driver = { .id_table = wm8971_i2c_id, }; -static int __init wm8971_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8971_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8971 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8971_modinit); - -static void __exit wm8971_exit(void) -{ - i2c_del_driver(&wm8971_i2c_driver); -} -module_exit(wm8971_exit); +module_i2c_driver(wm8971_i2c_driver); MODULE_DESCRIPTION("ASoC WM8971 driver"); MODULE_AUTHOR("Lab126"); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index d93c03f820c9..9a39511af52a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -659,23 +659,7 @@ static struct i2c_driver wm8974_i2c_driver = { .id_table = wm8974_i2c_id, }; -static int __init wm8974_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8974_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8974 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8974_modinit); - -static void __exit wm8974_exit(void) -{ - i2c_del_driver(&wm8974_i2c_driver); -} -module_exit(wm8974_exit); +module_i2c_driver(wm8974_i2c_driver); MODULE_DESCRIPTION("ASoC WM8974 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index a5be3adecf75..5421fd9fbcb5 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1105,23 +1105,7 @@ static struct i2c_driver wm8978_i2c_driver = { .id_table = wm8978_i2c_id, }; -static int __init wm8978_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8978_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8978 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8978_modinit); - -static void __exit wm8978_exit(void) -{ - i2c_del_driver(&wm8978_i2c_driver); -} -module_exit(wm8978_exit); +module_i2c_driver(wm8978_i2c_driver); MODULE_DESCRIPTION("ASoC WM8978 codec driver"); MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski@gmx.de>"); diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 9ac31ba9b82e..b9dbfebbda13 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1400,23 +1400,7 @@ static struct i2c_driver wm8991_i2c_driver = { .id_table = wm8991_i2c_id, }; -static int __init wm8991_modinit(void) -{ - int ret; - ret = i2c_add_driver(&wm8991_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n", - ret); - } - return 0; -} -module_init(wm8991_modinit); - -static void __exit wm8991_exit(void) -{ - i2c_del_driver(&wm8991_i2c_driver); -} -module_exit(wm8991_exit); +module_i2c_driver(wm8991_i2c_driver); MODULE_DESCRIPTION("ASoC WM8991 driver"); MODULE_AUTHOR("Graeme Gregory"); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 9fd80d688979..94737a30716b 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1520,6 +1520,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->pdata.lineout2fb, wm8993->pdata.jd_scthr, wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_delay, + wm8993->pdata.micbias2_delay, wm8993->pdata.micbias1_lvl, wm8993->pdata.micbias2_lvl); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bb62f4b3d563..b74df52d2820 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -671,6 +671,18 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const struct snd_kcontrol_new wm8994_drc_controls[] = { +SND_SOC_BYTES_MASK("AIF1.1 DRC", WM8994_AIF1_DRC1_1, 5, + WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF1.2 DRC", WM8994_AIF1_DRC2_1, 5, + WM8994_AIF1DAC2_DRC_ENA | WM8994_AIF1ADC2L_DRC_ENA | + WM8994_AIF1ADC2R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF2 DRC", WM8994_AIF2_DRC_1, 5, + WM8994_AIF2DAC_DRC_ENA | WM8994_AIF2ADCL_DRC_ENA | + WM8994_AIF2ADCR_DRC_ENA), +}; + static const char *wm8958_ng_text[] = { "30ms", "125ms", "250ms", "500ms", }; @@ -789,11 +801,27 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: return configure_clock(codec); + case SND_SOC_DAPM_POST_PMU: + /* + * JACKDET won't run until we start the clock and it + * only reports deltas, make sure we notify the state + * up the stack on startup. Use a *very* generous + * timeout for paranoia, there's no urgency and we + * don't want false reports. + */ + if (wm8994->jackdet && !wm8994->clk_has_run) { + schedule_delayed_work(&wm8994->jackdet_bootstrap, + msecs_to_jiffies(1000)); + wm8994->clk_has_run = true; + } + break; + case SND_SOC_DAPM_POST_PMD: configure_clock(codec); break; @@ -1632,7 +1660,8 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0), @@ -2102,6 +2131,10 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, case WM8994_FLL_SRC_LRCLK: case WM8994_FLL_SRC_BCLK: break; + case WM8994_FLL_SRC_INTERNAL: + freq_in = 12000000; + freq_out = 12000000; + break; default: return -EINVAL; } @@ -2161,12 +2194,14 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset, WM8994_FLL1_N_MASK, - fll.n << WM8994_FLL1_N_SHIFT); + fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, - WM8958_FLL1_BYP | + WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, + ((src == WM8994_FLL_SRC_INTERNAL) + << WM8994_FLL1_FRC_NCO_SHIFT) | (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | (src - 1)); @@ -2192,13 +2227,16 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } + reg = WM8994_FLL1_ENA; + if (fll.k) - reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC; - else - reg = WM8994_FLL1_ENA; + reg |= WM8994_FLL1_FRAC; + if (src == WM8994_FLL_SRC_INTERNAL) + reg |= WM8994_FLL1_OSC_ENA; + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, - WM8994_FLL1_ENA | WM8994_FLL1_FRAC, - reg); + WM8994_FLL1_ENA | WM8994_FLL1_OSC_ENA | + WM8994_FLL1_FRAC, reg); if (wm8994->fll_locked_irq) { timeout = wait_for_completion_timeout(&wm8994->fll_locked[id], @@ -2649,7 +2687,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 2; + bclk_rate = params_rate(params) * 4; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; @@ -3027,7 +3065,7 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) { - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; struct wm8994_pdata *pdata = wm8994->pdata; struct snd_kcontrol_new controls[] = { SOC_ENUM_EXT("AIF1.1 EQ Mode", @@ -3085,16 +3123,16 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, ARRAY_SIZE(controls)); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add ReTune Mobile controls: %d\n", ret); } static void wm8994_handle_pdata(struct wm8994_priv *wm8994) { - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; struct wm8994_pdata *pdata = wm8994->pdata; int ret, i; @@ -3107,6 +3145,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) pdata->lineout2fb, pdata->jd_scthr, pdata->jd_thr, + pdata->micb1_delay, + pdata->micb2_delay, pdata->micbias1_lvl, pdata->micbias2_lvl); @@ -3123,10 +3163,10 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) }; /* We need an array of texts for the enum API */ - wm8994->drc_texts = devm_kzalloc(wm8994->codec->dev, + wm8994->drc_texts = devm_kzalloc(wm8994->hubs.codec->dev, sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); if (!wm8994->drc_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d DRC config texts\n", pdata->num_drc_cfgs); return; @@ -3138,23 +3178,28 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) wm8994->drc_enum.max = pdata->num_drc_cfgs; wm8994->drc_enum.texts = wm8994->drc_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, ARRAY_SIZE(controls)); - if (ret != 0) - dev_err(wm8994->codec->dev, - "Failed to add DRC mode controls: %d\n", ret); - for (i = 0; i < WM8994_NUM_DRC; i++) wm8994_set_drc(codec, i); + } else { + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + wm8994_drc_controls, + ARRAY_SIZE(wm8994_drc_controls)); } + if (ret != 0) + dev_err(wm8994->hubs.codec->dev, + "Failed to add DRC mode controls: %d\n", ret); + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", pdata->num_retune_mobile_cfgs); if (pdata->num_retune_mobile_cfgs) wm8994_handle_retune_mobile_pdata(wm8994); else - snd_soc_add_codec_controls(wm8994->codec, wm8994_eq_controls, + snd_soc_add_codec_controls(wm8994->hubs.codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { @@ -3236,6 +3281,12 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, reg); + /* enable MICDET and MICSHRT deboune */ + snd_soc_update_bits(codec, WM8994_IRQ_DEBOUNCE, + WM8994_MIC1_DET_DB_MASK | WM8994_MIC1_SHRT_DB_MASK | + WM8994_MIC2_DET_DB_MASK | WM8994_MIC2_SHRT_DB_MASK, + WM8994_MIC1_DET_DB | WM8994_MIC1_SHRT_DB); + snd_soc_dapm_sync(&codec->dapm); return 0; @@ -3253,10 +3304,13 @@ static void wm8994_mic_work(struct work_struct *work) int ret; int report; + pm_runtime_get_sync(dev); + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); if (ret < 0) { dev_err(dev, "Failed to read microphone status: %d\n", ret); + pm_runtime_put(dev); return; } @@ -3299,12 +3353,14 @@ static void wm8994_mic_work(struct work_struct *work) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); + + pm_runtime_put(dev); } static irqreturn_t wm8994_mic_irq(int irq, void *data) { struct wm8994_priv *priv = data; - struct snd_soc_codec *codec = priv->codec; + struct snd_soc_codec *codec = priv->hubs.codec; #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); @@ -3340,7 +3396,7 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_jack_report(wm8994->micdet[0].jack, 0, wm8994->btn_mask | - SND_JACK_HEADSET); + SND_JACK_HEADSET); } return; } @@ -3417,16 +3473,19 @@ static void wm8958_default_micdet(u16 status, void *data) static irqreturn_t wm1811_jackdet_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; int reg; bool present; + pm_runtime_get_sync(codec->dev); + mutex_lock(&wm8994->accdet_lock); reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); if (reg < 0) { dev_err(codec->dev, "Failed to read jack status: %d\n", reg); mutex_unlock(&wm8994->accdet_lock); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3491,9 +3550,22 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + /* Since we only report deltas force an update, ensures we + * avoid bootstrapping issues with the core. */ + snd_soc_jack_report(wm8994->micdet[0].jack, 0, 0); + + pm_runtime_put(codec->dev); return IRQ_HANDLED; } +static void wm1811_jackdet_bootstrap(struct work_struct *work) +{ + struct wm8994_priv *wm8994 = container_of(work, + struct wm8994_priv, + jackdet_bootstrap.work); + wm1811_jackdet_irq(0, wm8994); +} + /** * wm8958_mic_detect - Enable microphone detection via the WM8958 IRQ * @@ -3564,6 +3636,10 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, * otherwise jump straight to microphone detection. */ if (wm8994->jackdet) { + /* Disable debounce for the initial detect */ + snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, + WM1811_JACKDET_DB, 0); + snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); @@ -3591,7 +3667,7 @@ EXPORT_SYMBOL_GPL(wm8958_mic_detect); static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; int reg, count; /* @@ -3602,6 +3678,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + pm_runtime_get_sync(codec->dev); + /* We may occasionally read a detection without an impedence * range being provided - if that happens loop again. */ @@ -3612,6 +3690,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_err(codec->dev, "Failed to read mic detect status: %d\n", reg); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3639,6 +3718,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_warn(codec->dev, "Accessory detection with no callback\n"); out: + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -3677,15 +3757,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) unsigned int reg; int ret, i; - wm8994->codec = codec; + wm8994->hubs.codec = codec; codec->control_data = control->regmap; snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - wm8994->codec = codec; - mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); + INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, + wm1811_jackdet_bootstrap); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index d77e06f0a675..f142ec198db3 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -28,10 +28,11 @@ #define WM8994_FLL1 1 #define WM8994_FLL2 2 -#define WM8994_FLL_SRC_MCLK1 1 -#define WM8994_FLL_SRC_MCLK2 2 -#define WM8994_FLL_SRC_LRCLK 3 -#define WM8994_FLL_SRC_BCLK 4 +#define WM8994_FLL_SRC_MCLK1 1 +#define WM8994_FLL_SRC_MCLK2 2 +#define WM8994_FLL_SRC_LRCLK 3 +#define WM8994_FLL_SRC_BCLK 4 +#define WM8994_FLL_SRC_INTERNAL 5 enum wm8994_vmid_mode { WM8994_VMID_NORMAL, @@ -72,7 +73,6 @@ struct wm8994; struct wm8994_priv { struct wm_hubs_data hubs; struct wm8994 *wm8994; - struct snd_soc_codec *codec; int sysclk[2]; int sysclk_rate[2]; int mclk[2]; @@ -81,6 +81,7 @@ struct wm8994_priv { struct completion fll_locked[2]; bool fll_locked_irq; bool fll_byp; + bool clk_has_run; int vmid_refcount; int active_refcount; @@ -134,6 +135,7 @@ struct wm8994_priv { int btn_mask; bool jackdet; int jackdet_mode; + struct delayed_work jackdet_bootstrap; wm8958_micdet_cb jack_cb; void *jack_cb_data; diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 2c2346fdd637..c7ddc56175d1 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -695,17 +695,7 @@ static struct i2c_driver wm9090_i2c_driver = { .id_table = wm9090_id, }; -static int __init wm9090_init(void) -{ - return i2c_add_driver(&wm9090_i2c_driver); -} -module_init(wm9090_init); - -static void __exit wm9090_exit(void) -{ - i2c_del_driver(&wm9090_i2c_driver); -} -module_exit(wm9090_exit); +module_i2c_driver(wm9090_i2c_driver); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("WM9090 ASoC driver"); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 099e6ec32125..6aa1bf8c6897 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -132,8 +132,9 @@ SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1), SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), -SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), -SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1), +SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv), +SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1, + boost_tlv), SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), @@ -146,7 +147,7 @@ SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), -SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), +SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), @@ -619,6 +620,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); @@ -683,8 +685,8 @@ static int __devexit wm9712_remove(struct platform_device *pdev) static struct platform_driver wm9712_codec_driver = { .driver = { - .name = "wm9712-codec", - .owner = THIS_MODULE, + .name = "wm9712-codec", + .owner = THIS_MODULE, }, .probe = wm9712_probe, diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 3eb19fb71d17..d0b8a3287a85 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (wm9713 == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); + codec->control_data = wm9713; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 61baa48823cb..7a773a835b8e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -199,15 +199,56 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) list_add_tail(&cache->list, &hubs->dcs_cache); } +static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec, + u16 *reg_l, u16 *reg_r) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + u16 dcs_reg, reg; + + switch (hubs->dcs_readback_mode) { + case 2: + dcs_reg = WM8994_DC_SERVO_4E; + break; + case 1: + dcs_reg = WM8994_DC_SERVO_READBACK; + break; + default: + dcs_reg = WM8993_DC_SERVO_3; + break; + } + + /* Different chips in the family support different readback + * methods. + */ + switch (hubs->dcs_readback_mode) { + case 0: + *reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) + & WM8993_DCS_INTEG_CHAN_0_MASK; + *reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) + & WM8993_DCS_INTEG_CHAN_1_MASK; + break; + case 2: + case 1: + reg = snd_soc_read(codec, dcs_reg); + *reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; + *reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + break; + default: + WARN(1, "Unknown DCS readback method\n"); + return; + } +} + /* * Startup calibration of the DC servo */ -static void calibrate_dc_servo(struct snd_soc_codec *codec) +static void enable_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct wm_hubs_dcs_cache *cache; s8 offset; - u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; + u16 reg_l, reg_r, dcs_cfg, dcs_reg; switch (hubs->dcs_readback_mode) { case 2: @@ -245,27 +286,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) WM8993_DCS_TRIG_STARTUP_1); } - /* Different chips in the family support different readback - * methods. - */ - switch (hubs->dcs_readback_mode) { - case 0: - reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) - & WM8993_DCS_INTEG_CHAN_0_MASK; - reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) - & WM8993_DCS_INTEG_CHAN_1_MASK; - break; - case 2: - case 1: - reg = snd_soc_read(codec, dcs_reg); - reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) - >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; - reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; - break; - default: - WARN(1, "Unknown DCS readback method\n"); - return; - } + wm_hubs_read_dc_servo(codec, ®_l, ®_r); dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); @@ -276,12 +297,16 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) hubs->dcs_codes_l, hubs->dcs_codes_r); /* HPOUT1R */ - offset = reg_r; + offset = (s8)reg_r; + dev_dbg(codec->dev, "DCS right %d->%d\n", offset, + offset + hubs->dcs_codes_r); offset += hubs->dcs_codes_r; dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1L */ - offset = reg_l; + offset = (s8)reg_l; + dev_dbg(codec->dev, "DCS left %d->%d\n", offset, + offset + hubs->dcs_codes_l); offset += hubs->dcs_codes_l; dcs_cfg |= (u8)offset; @@ -535,7 +560,7 @@ static int hp_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_TIMER_PERIOD_01_MASK, 0); - calibrate_dc_servo(codec); + enable_dc_servo(codec); reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT | WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT; @@ -619,6 +644,28 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + switch (w->shift) { + case WM8993_MICB1_ENA_SHIFT: + if (hubs->micb1_delay) + msleep(hubs->micb1_delay); + break; + case WM8993_MICB2_ENA_SHIFT: + if (hubs->micb2_delay) + msleep(hubs->micb2_delay); + break; + default: + return -EINVAL; + } + + return 0; +} + void wm_hubs_update_class_w(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); @@ -809,8 +856,10 @@ SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), SND_SOC_DAPM_INPUT("IN2RP:VXRP"), -SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), @@ -1112,6 +1161,8 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + hubs->codec = codec; + INIT_LIST_HEAD(&hubs->dcs_cache); init_completion(&hubs->dcs_done); @@ -1143,13 +1194,16 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, - int jd_scthr, int jd_thr, int micbias1_lvl, - int micbias2_lvl) + int jd_scthr, int jd_thr, + int micbias1_delay, int micbias2_delay, + int micbias1_lvl, int micbias2_lvl) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); hubs->lineout1_se = !lineout1_diff; hubs->lineout2_se = !lineout2_diff; + hubs->micb1_delay = micbias1_delay; + hubs->micb2_delay = micbias2_delay; if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index da2dc899ce6d..24c763df21f9 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -36,6 +36,9 @@ struct wm_hubs_data { struct list_head dcs_cache; bool (*check_class_w_digital)(struct snd_soc_codec *); + int micb1_delay; + int micb2_delay; + bool lineout1_se; bool lineout1n_ena; bool lineout1p_ena; @@ -46,6 +49,8 @@ struct wm_hubs_data { bool dcs_done_irq; struct completion dcs_done; + + struct snd_soc_codec *codec; }; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); @@ -54,6 +59,7 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, int jd_scthr, int jd_thr, + int micbias1_dly, int micbias2_dly, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 10a2d8c788b7..c80c20a89b11 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -24,7 +24,6 @@ #include <mach/asp.h> #include <mach/edma.h> -#include <mach/mux.h> #include "davinci-pcm.h" #include "davinci-i2s.h" diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bfc8190..d919fb8de7a3 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -21,7 +21,7 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/io.h> -#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <sound/core.h> #include <sound/pcm.h> @@ -776,20 +776,17 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!dev->clk_active) { - clk_enable(dev->clk); - dev->clk_active = 1; - } + ret = pm_runtime_get_sync(dev->dev); + if (IS_ERR_VALUE(ret)) + dev_err(dev->dev, "pm_runtime_get_sync() failed\n"); davinci_mcasp_start(dev, substream->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: davinci_mcasp_stop(dev, substream->stream); - if (dev->clk_active) { - clk_disable(dev->clk); - dev->clk_active = 0; - } - + ret = pm_runtime_put_sync(dev->dev); + if (IS_ERR_VALUE(ret)) + dev_err(dev->dev, "pm_runtime_put_sync() failed\n"); break; case SNDRV_PCM_TRIGGER_STOP: @@ -886,12 +883,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } pdata = pdev->dev.platform_data; - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return -ENODEV; + pm_runtime_enable(&pdev->dev); - clk_enable(dev->clk); - dev->clk_active = 1; + ret = pm_runtime_get_sync(&pdev->dev); + if (IS_ERR_VALUE(ret)) { + dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); + return ret; + } dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!dev->base) { @@ -908,6 +906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; + dev->dev = &pdev->dev; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; @@ -949,19 +948,18 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err_release_clk: - clk_disable(dev->clk); - clk_put(dev->clk); + pm_runtime_put_sync(&pdev->dev); + pm_runtime_disable(&pdev->dev); return ret; } static int davinci_mcasp_remove(struct platform_device *pdev) { - struct davinci_audio_dev *dev = dev_get_drvdata(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - clk_disable(dev->clk); - clk_put(dev->clk); - dev->clk = NULL; + + pm_runtime_put_sync(&pdev->dev); + pm_runtime_disable(&pdev->dev); return 0; } diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 4681acc63606..51479f9ee909 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,8 @@ struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; - struct clk *clk; + struct device *dev; unsigned int codec_fmt; - u8 clk_active; /* McASP specific data */ int tdm_slots; diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c7cb1c..3c520c46fa4a 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -523,7 +523,7 @@ static int imx_ssi_probe(struct platform_device *pdev) int ret = 0; struct snd_soc_dai_driver *dai; - ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); + ssi = devm_kzalloc(&pdev->dev, sizeof(*ssi), GFP_KERNEL); if (!ssi) return -ENOMEM; dev_set_drvdata(&pdev->dev, ssi); @@ -536,7 +536,7 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->irq = platform_get_irq(pdev, 0); - ssi->clk = clk_get(&pdev->dev, NULL); + ssi->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi->clk)) { ret = PTR_ERR(ssi->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", @@ -551,23 +551,18 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_get_resource; } - if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { - dev_err(&pdev->dev, "request_mem_region failed\n"); - ret = -EBUSY; - goto failed_get_resource; - } - - ssi->base = ioremap(res->start, resource_size(res)); + ssi->base = devm_request_and_ioremap(&pdev->dev, res); if (!ssi->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; - goto failed_ioremap; + goto failed_register; } if (ssi->flags & IMX_SSI_USE_AC97) { if (ac97_ssi) { + dev_err(&pdev->dev, "AC'97 SSI already registered\n"); ret = -EBUSY; - goto failed_ac97; + goto failed_register; } ac97_ssi = ssi; setup_channel_to_ac97(ssi); @@ -636,15 +631,10 @@ failed_pdev_fiq_add: failed_pdev_fiq_alloc: snd_soc_unregister_dai(&pdev->dev); failed_register: -failed_ac97: - iounmap(ssi->base); -failed_ioremap: release_mem_region(res->start, resource_size(res)); failed_get_resource: clk_disable_unprepare(ssi->clk); - clk_put(ssi->clk); failed_clk: - kfree(ssi); return ret; } @@ -662,11 +652,8 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - iounmap(ssi->base); release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); - clk_put(ssi->clk); - kfree(ssi); return 0; } diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 2937e54da49e..2cc7782714b5 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -318,6 +318,15 @@ static struct snd_soc_dai_link mfld_msic_dailink[] = { .platform_name = "sst-platform", .init = NULL, }, + { + .name = "Medfield Compress", + .stream_name = "Speaker", + .cpu_dai_name = "Compress-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, }; /* SoC card */ diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/mid-x86/sst_dsp.h new file mode 100644 index 000000000000..0fce1de284ff --- /dev/null +++ b/sound/soc/mid-x86/sst_dsp.h @@ -0,0 +1,134 @@ +#ifndef __SST_DSP_H__ +#define __SST_DSP_H__ +/* + * sst_dsp.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-12 Intel Corporation + * Authors: Vinod Koul <vinod.koul@linux.intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +enum sst_codec_types { + /* AUDIO/MUSIC CODEC Type Definitions */ + SST_CODEC_TYPE_UNKNOWN = 0, + SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ + SST_CODEC_TYPE_MP3, + SST_CODEC_TYPE_MP24, + SST_CODEC_TYPE_AAC, + SST_CODEC_TYPE_AACP, + SST_CODEC_TYPE_eAACP, +}; + +enum stream_type { + SST_STREAM_TYPE_NONE = 0, + SST_STREAM_TYPE_MUSIC = 1, +}; + +struct snd_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u8 use_offload_path; + u8 reserved2; + u16 reserved3; + u8 channel_map[8]; +} __packed; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 crc_check; /* crc_check - disable (0) or enable (1) */ + u8 reserved1; /* unused*/ + u16 reserved2; /* Unused */ +} __packed; + +#define AAC_BIT_STREAM_ADTS 0 +#define AAC_BIT_STREAM_ADIF 1 +#define AAC_BIT_STREAM_RAW 2 + +/* AAC Music Parameters Message */ +struct snd_aac_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo*/ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */ + u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ + u16 reser2; + u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ + u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ + u8 reser1; + u16 reser3; +} __packed; + +/* WMA Music Parameters Message */ +struct snd_wma_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 brate; /* Use the hard coded value. */ + u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + u32 channel_mask; /* Channel Mask */ + u16 format_tag; /* Format Tag */ + u16 block_align; /* packet size */ + u16 wma_encode_opt;/* Encoder option */ + u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */ + u8 reserved; /* reserved */ +} __packed; + +/* Codec params struture */ +union snd_sst_codec_params { + struct snd_pcm_params pcm_params; + struct snd_mp3_params mp3_params; + struct snd_aac_params aac_params; + struct snd_wma_params wma_params; +} __packed; + +/* Address and size info of a frame buffer */ +struct sst_address_info { + u32 addr; /* Address at IA */ + u32 size; /* Size of the buffer */ +}; + +struct snd_sst_alloc_params_ext { + struct sst_address_info ring_buf_info[8]; + u8 sg_count; + u8 reserved; + u16 reserved2; + u32 frag_size; /*Number of samples after which period elapsed + message is sent valid only if path = 0*/ +} __packed; + +struct snd_sst_stream_params { + union snd_sst_codec_params uc; +} __packed; + +struct snd_sst_params { + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct snd_sst_stream_params sparams; + struct snd_sst_alloc_params_ext aparams; +}; + +#endif /* __SST_DSP_H__ */ diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index d34563b12c3b..a263cbed8624 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -1,7 +1,7 @@ /* * sst_platform.c - Intel MID Platform driver * - * Copyright (C) 2010 Intel Corp + * Copyright (C) 2010-2012 Intel Corp * Author: Vinod Koul <vinod.koul@intel.com> * Author: Harsha Priya <priya.harsha@intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -32,6 +32,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/compress_driver.h> #include "sst_platform.h" static struct sst_device *sst; @@ -152,6 +153,16 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE, }, }, +{ + .name = "Compress-cpu-dai", + .compress_dai = 1, + .playback = { + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, }; /* helper functions */ @@ -463,8 +474,199 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) } return retval; } + +/* compress stream operations */ +static void sst_compr_fragment_elapsed(void *arg) +{ + struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; + + pr_debug("fragment elapsed by driver\n"); + if (cstream) + snd_compr_fragment_elapsed(cstream); +} + +static int sst_platform_compr_open(struct snd_compr_stream *cstream) +{ + + int ret_val = 0; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sst_runtime_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + if (!sst || !try_module_get(sst->dev->driver->owner)) { + pr_err("no device available to run\n"); + ret_val = -ENODEV; + goto out_ops; + } + stream->compr_ops = sst->compr_ops; + + stream->id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + runtime->private_data = stream; + return 0; +out_ops: + kfree(stream); + return ret_val; +} + +static int sst_platform_compr_free(struct snd_compr_stream *cstream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + stream = cstream->runtime->private_data; + /*need to check*/ + str_id = stream->id; + if (str_id) + ret_val = stream->compr_ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + pr_debug("%s: %d\n", __func__, ret_val); + return 0; +} + +static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct sst_runtime_stream *stream; + int retval; + struct snd_sst_params str_params; + struct sst_compress_cb cb; + + stream = cstream->runtime->private_data; + /* construct fw structure for this*/ + memset(&str_params, 0, sizeof(str_params)); + + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.stream_type = SST_STREAM_TYPE_MUSIC; + str_params.device_type = SND_SST_DEVICE_COMPRESS; + + switch (params->codec.id) { + case SND_AUDIOCODEC_MP3: { + str_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; + break; + } + + case SND_AUDIOCODEC_AAC: { + str_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.aac_params.pcm_wd_sz = 16; + if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_ADTS; + else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_RAW; + else { + pr_err("Undefined format%d\n", params->codec.format); + return -EINVAL; + } + str_params.sparams.uc.aac_params.externalsr = + params->codec.sample_rate; + break; + } + + default: + pr_err("codec not supported, id =%d\n", params->codec.id); + return -EINVAL; + } + + str_params.aparams.ring_buf_info[0].addr = + virt_to_phys(cstream->runtime->buffer); + str_params.aparams.ring_buf_info[0].size = + cstream->runtime->buffer_size; + str_params.aparams.sg_count = 1; + str_params.aparams.frag_size = cstream->runtime->fragment_size; + + cb.param = cstream; + cb.compr_cb = sst_compr_fragment_elapsed; + + retval = stream->compr_ops->open(&str_params, &cb); + if (retval < 0) { + pr_err("stream allocation failed %d\n", retval); + return retval; + } + + stream->id = retval; + return 0; +} + +static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->control(cmd, stream->id); +} + +static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->tstamp(stream->id, tstamp); + tstamp->byte_offset = tstamp->copied_total % + (u32)cstream->runtime->buffer_size; + pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); + return 0; +} + +static int sst_platform_compr_ack(struct snd_compr_stream *cstream, + size_t bytes) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->bytes_written += bytes; + + return 0; +} + +static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_caps(caps); +} + +static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_codec_caps(codec); +} + +static struct snd_compr_ops sst_platform_compr_ops = { + + .open = sst_platform_compr_open, + .free = sst_platform_compr_free, + .set_params = sst_platform_compr_set_params, + .trigger = sst_platform_compr_trigger, + .pointer = sst_platform_compr_pointer, + .ack = sst_platform_compr_ack, + .get_caps = sst_platform_compr_get_caps, + .get_codec_caps = sst_platform_compr_get_codec_caps, +}; + static struct snd_soc_platform_driver sst_soc_platform_drv = { .ops = &sst_platform_ops, + .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, .pcm_free = sst_pcm_free, }; diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h index f04f4f72daa0..d61c5d514ffa 100644 --- a/sound/soc/mid-x86/sst_platform.h +++ b/sound/soc/mid-x86/sst_platform.h @@ -27,6 +27,8 @@ #ifndef __SST_PLATFORMDRV_H__ #define __SST_PLATFORMDRV_H__ +#include "sst_dsp.h" + #define SST_MONO 1 #define SST_STEREO 2 #define SST_MAX_CAP 5 @@ -42,7 +44,6 @@ #define SST_MIN_PERIODS 2 #define SST_MAX_PERIODS (1024*2) #define SST_FIFO_SIZE 0 -#define SST_CODEC_TYPE_PCM 1 struct pcm_stream_info { int str_id; @@ -83,6 +84,7 @@ enum sst_audio_device_type { SND_SST_DEVICE_VIBRA, SND_SST_DEVICE_HAPTIC, SND_SST_DEVICE_CAPTURE, + SND_SST_DEVICE_COMPRESS, }; /* PCM Parameters */ @@ -107,6 +109,24 @@ struct sst_stream_params { struct sst_pcm_params sparams; }; +struct sst_compress_cb { + void *param; + void (*compr_cb)(void *param); +}; + +struct compress_sst_ops { + const char *name; + int (*open) (struct snd_sst_params *str_params, + struct sst_compress_cb *cb); + int (*control) (unsigned int cmd, unsigned int str_id); + int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); + int (*ack) (unsigned int str_id, unsigned long bytes); + int (*close) (unsigned int str_id); + int (*get_caps) (struct snd_compr_caps *caps); + int (*get_codec_caps) (struct snd_compr_codec_caps *codec); + +}; + struct sst_ops { int (*open) (struct sst_stream_params *str_param); int (*device_control) (int cmd, void *arg); @@ -115,8 +135,11 @@ struct sst_ops { struct sst_runtime_stream { int stream_status; + unsigned int id; + size_t bytes_written; struct pcm_stream_info stream_info; struct sst_ops *ops; + struct compress_sst_ops *compr_ops; spinlock_t status_lock; }; @@ -124,6 +147,7 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct compress_sst_ops *compr_ops; }; int sst_register_dsp(struct sst_device *sst); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aba71bfa33b1..aa037b292f3d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; u32 scr, stat; int ret; + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + /* mclk should already be set */ if (!saif->mclk && saif->mclk_in_use) { dev_err(cpu_dai->dev, "set mclk first\n"); @@ -420,6 +425,25 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, return ret; } + /* prepare clk in hw_param, enable in trigger */ + clk_prepare(saif->clk); + if (saif != master_saif) { + /* + * Set an initial clock rate for the saif internal logic to work + * properly. This is important when working in EXTMASTER mode + * that uses the other saif's BITCLK&LRCLK but it still needs a + * basic clock which should be fast enough for the internal + * logic. + */ + clk_enable(saif->clk); + ret = clk_set_rate(saif->clk, 24000000); + clk_disable(saif->clk); + if (ret) + return ret; + + clk_prepare(master_saif->clk); + } + scr = __raw_readl(saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_WORD_LENGTH; @@ -680,7 +704,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->clk = clk_get(&pdev->dev, NULL); + saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", @@ -693,8 +717,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; - goto failed_get_resource; + return -ENODEV; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -707,7 +730,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) &saif->dma_param.chan_num); if (ret) { dev_err(&pdev->dev, "failed to get dma channel\n"); - goto failed_get_resource; + return ret; } } else { saif->dma_param.chan_num = dmares->start; @@ -718,7 +741,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } saif->dev = &pdev->dev; @@ -726,7 +749,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_resource; + return ret; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -734,7 +757,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } platform_set_drvdata(pdev, saif); @@ -742,7 +765,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_get_resource; + return ret; } ret = mxs_pcm_platform_register(&pdev->dev); @@ -755,19 +778,14 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_get_resource: - clk_put(saif->clk); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct mxs_saif *saif = platform_get_drvdata(pdev); - mxs_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - clk_put(saif->clk); return 0; } diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 9d93793d3077..be525dfe9faa 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -25,6 +25,7 @@ #include <linux/mfd/twl6040.h> #include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> +#include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,6 +44,8 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ + + struct platform_device *dmic_codec_dev; }; static int omap_abe_hw_params(struct snd_pcm_substream *substream, @@ -185,17 +188,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) int hs_trim; int ret = 0; - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - /* * Configure McPDM offset cancellation based on the HSOTRIM value from * twl6040. @@ -216,6 +208,24 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } + /* + * NULL pdata means we booted with DT. In this case the routing is + * provided and the card is fully routed, no need to mark pins. + */ + if (!pdata) + return ret; + + /* Disable not connected paths if not used */ + twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); + twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); + twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); + return ret; } @@ -270,52 +280,116 @@ static struct snd_soc_card omap_abe_card = { static __devinit int omap_abe_probe(struct platform_device *pdev) { struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); + struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; struct abe_twl6040 *priv; int num_links = 0; - int ret; + int ret = 0; card->dev = &pdev->dev; - if (!pdata) { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; - } - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; - if (pdata->card_name) { - card->name = pdata->card_name; + priv->dmic_codec_dev = ERR_PTR(-EINVAL); + + if (node) { + struct device_node *dai_node; + + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + ret = snd_soc_of_parse_audio_routing(card, + "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, + "Error while parsing DAPM routing\n"); + return ret; + } + + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; + + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; + + priv->dmic_codec_dev = platform_device_register_simple( + "dmic-codec", -1, NULL, 0); + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, + "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); + } + } else { + num_links = 1; + } + + of_property_read_u32(node, "ti,jack-detection", + &priv->jack_detection); + of_property_read_u32(node, "ti,mclk-freq", + &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; + } + + omap_abe_card.fully_routed = 1; + } else if (pdata) { + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + if (pdata->has_dmic) + num_links = 2; + else + num_links = 1; + + priv->jack_detection = pdata->jack_detection; + priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Card name is not provided\n"); + dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; - if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - return -ENODEV; + ret = -ENODEV; + goto err_unregister; } - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - card->dai_link = abe_twl6040_dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + goto err_unregister; + } + + return 0; + +err_unregister: + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -323,17 +397,28 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) static int __devexit omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); + return 0; } +static const struct of_device_id omap_abe_of_match[] = { + {.compatible = "ti,abe-twl6040", }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_abe_of_match); + static struct platform_driver omap_abe_driver = { .driver = { .name = "omap-abe-twl6040", .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, .remove = __devexit_p(omap_abe_remove), diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2e1750e2ab31..d6de066ad7ea 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -860,3 +860,4 @@ module_platform_driver(asoc_mcbsp_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-mcbsp"); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 2c66e2498a45..f7babb374a37 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -445,9 +445,8 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { struct omap_mcpdm *mcpdm; struct resource *res; - int ret = 0; - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL); if (!mcpdm) return -ENOMEM; @@ -456,55 +455,30 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) mutex_init(&mcpdm->mutex); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_res; - } + if (res == NULL) + return -ENOMEM; - if (!request_mem_region(res->start, resource_size(res), "McPDM")) { - ret = -EBUSY; - goto err_res; - } + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), "McPDM")) + return -EBUSY; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_iomap; - } + mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!mcpdm->io_base) + return -ENOMEM; mcpdm->irq = platform_get_irq(pdev, 0); - if (mcpdm->irq < 0) { - ret = mcpdm->irq; - goto err_irq; - } + if (mcpdm->irq < 0) + return mcpdm->irq; mcpdm->dev = &pdev->dev; - ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (!ret) - return 0; - -err_irq: - iounmap(mcpdm->io_base); -err_iomap: - release_mem_region(res->start, resource_size(res)); -err_res: - kfree(mcpdm); - return ret; + return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); } static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); - struct resource *res; - snd_soc_unregister_dai(&pdev->dev); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - iounmap(mcpdm->io_base); - release_mem_region(res->start, resource_size(res)); - - kfree(mcpdm); return 0; } diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5a649da9122a..f0feb06615f8 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-pcm-audio"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fe3995ce9b38..fb5600083612 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -199,6 +199,14 @@ config SND_SOC_TOBERMORY select SND_SAMSUNG_I2S select SND_SOC_WM8962 +config SND_SOC_BELLS + tristate "Audio support for Wolfson Bells" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM5102 + select SND_SOC_WM5110 + select SND_SOC_WM9081 + config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 9d03beb40c86..709f6059ad67 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -42,6 +42,7 @@ snd-soc-speyside-objs := speyside.o snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o +snd-soc-bells-objs := bells.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -65,3 +66,4 @@ obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o +obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c new file mode 100644 index 000000000000..5dc10dfc0d42 --- /dev/null +++ b/sound/soc/samsung/bells.c @@ -0,0 +1,346 @@ +/* + * Bells audio support + * + * Copyright 2012 Wolfson Microelectronics + * + * 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 <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/gpio.h> +#include <linux/module.h> + +#include "../codecs/wm5102.h" +#include "../codecs/wm9081.h" + +/* + * 44.1kHz based clocks for the SYSCLK domain, use a very high clock + * to allow all the DSP functionality to be enabled if desired. + */ +#define SYSCLK_RATE (44100 * 1024) + +/* 48kHz based clocks for the ASYNC domain */ +#define ASYNCCLK_RATE (48000 * 512) + +/* BCLK2 is fixed at this currently */ +#define BCLK2_RATE (64 * 8000) + +/* + * Expect a 24.576MHz crystal if one is fitted (the driver will function + * if this is not fitted). + */ +#define MCLK_RATE 24576000 + +#define WM9081_AUDIO_RATE 44100 +#define WM9081_MCLK_RATE (WM9081_AUDIO_RATE * 256) + +static int bells_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, + ARIZONA_FLL_SRC_MCLK1, + MCLK_RATE, + SYSCLK_RATE); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, + ARIZONA_FLL_SRC_AIF2BCLK, + BCLK2_RATE, + ASYNCCLK_RATE); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + } + break; + + default: + break; + } + + return 0; +} + +static int bells_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int bells_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + struct snd_soc_dai *aif3_dai = card->rtd[2].cpu_dai; + struct snd_soc_dai *wm9081_dai = card->rtd[2].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0); + if (ret != 0) { + dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); + if (ret != 0) { + dev_err(aif2_dai->dev, "Failed to set AIF2 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0); + if (ret != 0) { + dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK, + ARIZONA_CLK_SRC_FLL1, SYSCLK_RATE, + SND_SOC_CLOCK_IN); + if (ret != 0) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_OPCLK, 0, + WM9081_MCLK_RATE, SND_SOC_CLOCK_OUT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set OPCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK, + ARIZONA_CLK_SRC_FLL2, ASYNCCLK_RATE, + SND_SOC_CLOCK_IN); + if (ret != 0) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(wm9081_dai->codec, WM9081_SYSCLK_MCLK, + 0, WM9081_MCLK_RATE, 0); + if (ret != 0) { + dev_err(wm9081_dai->dev, "Failed to set MCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + +static const struct snd_soc_pcm_stream sub_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = WM9081_AUDIO_RATE, + .rate_max = WM9081_AUDIO_RATE, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_dai_link bells_dai_wm5102[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5102-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5102-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5102-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, + { + .name = "Sub", + .stream_name = "Sub", + .cpu_dai_name = "wm5102-aif3", + .codec_dai_name = "wm9081-hifi", + .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .params = &sub_params, + }, +}; + +static struct snd_soc_dai_link bells_dai_wm5110[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5110-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5110-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5110-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, + { + .name = "Sub", + .stream_name = "Sub", + .cpu_dai_name = "wm5102-aif3", + .codec_dai_name = "wm9081-hifi", + .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .params = &sub_params, + }, +}; + +static struct snd_soc_codec_conf bells_codec_conf[] = { + { + .dev_name = "wm9081.1-006c", + .name_prefix = "Sub", + }, +}; + +static struct snd_soc_dapm_route bells_routes[] = { + { "Sub CLK_SYS", NULL, "OPCLK" }, +}; + +static struct snd_soc_card bells_cards[] = { + { + .name = "Bells WM5102", + .owner = THIS_MODULE, + .dai_link = bells_dai_wm5102, + .num_links = ARRAY_SIZE(bells_dai_wm5102), + .codec_conf = bells_codec_conf, + .num_configs = ARRAY_SIZE(bells_codec_conf), + + .late_probe = bells_late_probe, + + .dapm_routes = bells_routes, + .num_dapm_routes = ARRAY_SIZE(bells_routes), + + .set_bias_level = bells_set_bias_level, + .set_bias_level_post = bells_set_bias_level_post, + }, + { + .name = "Bells WM5110", + .owner = THIS_MODULE, + .dai_link = bells_dai_wm5110, + .num_links = ARRAY_SIZE(bells_dai_wm5110), + .codec_conf = bells_codec_conf, + .num_configs = ARRAY_SIZE(bells_codec_conf), + + .late_probe = bells_late_probe, + + .dapm_routes = bells_routes, + .num_dapm_routes = ARRAY_SIZE(bells_routes), + + .set_bias_level = bells_set_bias_level, + .set_bias_level_post = bells_set_bias_level_post, + }, +}; + + +static __devinit int bells_probe(struct platform_device *pdev) +{ + int ret; + + bells_cards[pdev->id].dev = &pdev->dev; + + ret = snd_soc_register_card(&bells_cards[pdev->id]); + if (ret) { + dev_err(&pdev->dev, + "snd_soc_register_card(%s) failed: %d\n", + bells_cards[pdev->id].name, ret); + return ret; + } + + return 0; +} + +static int __devexit bells_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&bells_cards[pdev->id]); + + return 0; +} + +static struct platform_driver bells_driver = { + .driver = { + .name = "bells", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bells_probe, + .remove = __devexit_p(bells_remove), +}; + +module_platform_driver(bells_driver); + +MODULE_DESCRIPTION("Bells audio support"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bells"); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c new file mode 100644 index 000000000000..967d0e173e1b --- /dev/null +++ b/sound/soc/soc-compress.c @@ -0,0 +1,294 @@ +/* + * soc-compress.c -- ALSA SoC Compress + * + * Copyright (C) 2012 Intel Corp. + * + * Authors: Namarta Kohli <namartax.kohli@intel.com> + * Ramesh Babu K V <ramesh.babu@linux.intel.com> + * Vinod Koul <vinod.koul@linux.intel.com> + * + * 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 <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/compress_params.h> +#include <sound/compress_driver.h> +#include <sound/soc.h> +#include <sound/initval.h> + +static int soc_compr_open(struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->open) { + ret = platform->driver->compr_ops->open(cstream); + if (ret < 0) { + pr_err("compress asoc: can't open platform %s\n", platform->name); + goto out; + } + } + + if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->startup) { + ret = rtd->dai_link->compr_ops->startup(cstream); + if (ret < 0) { + pr_err("compress asoc: %s startup failed\n", rtd->dai_link->name); + goto machine_err; + } + } + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + + cpu_dai->active++; + codec_dai->active++; + rtd->codec->active++; + + return 0; + +machine_err: + if (platform->driver->compr_ops && platform->driver->compr_ops->free) + platform->driver->compr_ops->free(cstream); +out: + return ret; +} + +static int soc_compr_free(struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + snd_soc_dai_digital_mute(codec_dai, 1); + + cpu_dai->active--; + codec_dai->active--; + codec->active--; + + if (!cpu_dai->active) + cpu_dai->rate = 0; + + if (!codec_dai->active) + codec_dai->rate = 0; + + + if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->shutdown) + rtd->dai_link->compr_ops->shutdown(cstream); + + if (platform->driver->compr_ops && platform->driver->compr_ops->free) + platform->driver->compr_ops->free(cstream); + cpu_dai->runtime = NULL; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + if (!rtd->pmdown_time || codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time) { + snd_soc_dapm_stream_event(rtd, + SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_STOP); + } else + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } else { + /* capture streams can be powered down now */ + snd_soc_dapm_stream_event(rtd, + SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_STOP); + } + + return 0; +} + +static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->trigger) { + ret = platform->driver->compr_ops->trigger(cstream, cmd); + if (ret < 0) + return ret; + } + + if (cmd == SNDRV_PCM_TRIGGER_START) + snd_soc_dai_digital_mute(codec_dai, 0); + else if (cmd == SNDRV_PCM_TRIGGER_STOP) + snd_soc_dai_digital_mute(codec_dai, 1); + + return ret; +} + +static int soc_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + /* first we call set_params for the platform driver + * this should configure the soc side + * if the machine has compressed ops then we call that as well + * expectation is that platform and machine will configure everything + * for this compress path, like configuring pcm port for codec + */ + if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { + ret = platform->driver->compr_ops->set_params(cstream, params); + if (ret < 0) + return ret; + } + + if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { + ret = rtd->dai_link->compr_ops->set_params(cstream); + if (ret < 0) + return ret; + } + + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + + return ret; +} + +static int soc_compr_get_params(struct snd_compr_stream *cstream, + struct snd_codec *params) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_params) + ret = platform->driver->compr_ops->get_params(cstream, params); + + return ret; +} + +static int soc_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_caps) + ret = platform->driver->compr_ops->get_caps(cstream, caps); + + return ret; +} + +static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_codec_caps) + ret = platform->driver->compr_ops->get_codec_caps(cstream, codec); + + return ret; +} + +static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->ack) + ret = platform->driver->compr_ops->ack(cstream, bytes); + + return ret; +} + +static int soc_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + platform->driver->compr_ops->pointer(cstream, tstamp); + + return 0; +} + +/* ASoC Compress operations */ +static struct snd_compr_ops soc_compr_ops = { + .open = soc_compr_open, + .free = soc_compr_free, + .set_params = soc_compr_set_params, + .get_params = soc_compr_get_params, + .trigger = soc_compr_trigger, + .pointer = soc_compr_pointer, + .ack = soc_compr_ack, + .get_caps = soc_compr_get_caps, + .get_codec_caps = soc_compr_get_codec_caps +}; + +/* create a new compress */ +int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_compr *compr; + char new_name[64]; + int ret = 0, direction = 0; + + /* check client and interface hw capabilities */ + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + direction = SND_COMPRESS_PLAYBACK; + compr = kzalloc(sizeof(*compr), GFP_KERNEL); + if (compr == NULL) { + snd_printk(KERN_ERR "Cannot allocate compr\n"); + return -ENOMEM; + } + + compr->ops = &soc_compr_ops; + mutex_init(&compr->lock); + ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); + if (ret < 0) { + pr_err("compress asoc: can't create compress for codec %s\n", + codec->name); + kfree(compr); + return ret; + } + + rtd->compr = compr; + compr->private_data = rtd; + + printk(KERN_INFO "compress asoc: %s <-> %s mapping ok\n", codec_dai->name, + cpu_dai->name); + return ret; +} diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f219b2f7ee68..c7a00fd8cc66 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card, } /* If the driver didn't set I/O up try regmap */ - if (!codec->control_data) + if (!codec->write && dev_get_regmap(codec->dev, NULL)) snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (driver->controls) @@ -1388,37 +1388,48 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (ret < 0) pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); - if (!dai_link->params) { - /* create the pcm */ - ret = soc_new_pcm(rtd, num); + if (cpu_dai->driver->compress_dai) { + /*create compress_device"*/ + ret = soc_new_compress(rtd, num); if (ret < 0) { - pr_err("asoc: can't create pcm %s :%d\n", - dai_link->stream_name, ret); + pr_err("asoc: can't create compress %s\n", + dai_link->stream_name); return ret; } } else { - /* link the DAI widgets */ - play_w = codec_dai->playback_widget; - capture_w = cpu_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); + + if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); return ret; } - } + } else { + /* link the DAI widgets */ + play_w = codec_dai->playback_widget; + capture_w = cpu_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } - play_w = cpu_dai->playback_widget; - capture_w = codec_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, + play_w = cpu_dai->playback_widget; + capture_w = codec_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); - return ret; + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } } } } @@ -1814,7 +1825,6 @@ base_error: static int soc_probe(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - int ret = 0; /* * no card, so machine driver should be registering card @@ -1830,13 +1840,7 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - ret = snd_soc_register_card(card); - if (ret != 0) { - dev_err(&pdev->dev, "Failed to register card\n"); - return ret; - } - - return 0; + return snd_soc_register_card(card); } static int soc_cleanup_card_resources(struct snd_soc_card *card) @@ -3715,6 +3719,9 @@ int snd_soc_register_dai(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); @@ -3803,6 +3810,9 @@ int snd_soc_register_dais(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7428bb..2ca3c734a288 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -83,11 +83,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) jack->status &= ~mask; jack->status |= status & mask; - /* The DAPM sync is expensive enough to be worth skipping. - * However, empty mask means pin synchronization is desired. */ - if (mask && (jack->status == oldstatus)) - goto out; - trace_snd_soc_jack_notify(jack, status); list_for_each_entry(pin, &jack->pins, list) { @@ -109,7 +104,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_jack_report(jack->jack, jack->status); -out: mutex_unlock(&jack->mutex); } EXPORT_SYMBOL_GPL(snd_soc_jack_report); diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index d684df294c0c..e463529b38bb 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -177,7 +177,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) } alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (alc5632->gpio_hp_det == -ENODEV) + if (alc5632->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; ret = snd_soc_of_parse_card_name(card, "nvidia,model"); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0c5bb33d258e..d4f14e492341 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } else if (np) { pdata->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios", 0); - if (pdata->gpio_spkr_en == -ENODEV) + if (pdata->gpio_spkr_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_mute = of_get_named_gpio(np, "nvidia,hp-mute-gpios", 0); - if (pdata->gpio_hp_mute == -ENODEV) + if (pdata->gpio_hp_mute == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (pdata->gpio_hp_det == -ENODEV) + if (pdata->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_int_mic_en = of_get_named_gpio(np, "nvidia,int-mic-en-gpios", 0); - if (pdata->gpio_int_mic_en == -ENODEV) + if (pdata->gpio_int_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_ext_mic_en = of_get_named_gpio(np, "nvidia,ext-mic-en-gpios", 0); - if (pdata->gpio_ext_mic_en == -ENODEV) + if (pdata->gpio_ext_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; } diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 62ac0285bfaf..772cb19d2fb3 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -21,7 +21,7 @@ #include <linux/mfd/dbx500-prcmu.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -760,6 +760,9 @@ static int __devinit ux500_msp_drv_probe(struct platform_device *pdev) drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp_i2s_drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + drvdata->fmt = 0; drvdata->slots = 1; drvdata->tx_mask = 0x01; diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index ee14d2dac2f5..36be11e47ad6 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -19,7 +19,7 @@ #include <linux/slab.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> @@ -673,6 +673,8 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); msp = *msp_p; + if (!msp) + return -ENOMEM; msp->id = platform_data->id; msp->dev = &pdev->dev; diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 7f71b4a0d4bc..2d9136da9865 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -17,7 +17,7 @@ #include <linux/platform_device.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #define MSP_INPUT_FREQ_APB 48000000 |