diff options
Diffstat (limited to 'sound')
45 files changed, 453 insertions, 407 deletions
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index c3f57a3fb1a5..40ebde2e1ab1 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -47,8 +47,8 @@ static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, /* We use the PCI APIs for now until the generic one gets fixed * enough or until we get some macio-specific versions */ - r->space = dma_zalloc_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, - r->size, &r->bus_addr, GFP_KERNEL); + r->space = dma_alloc_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, + r->size, &r->bus_addr, GFP_KERNEL); if (!r->space) return -ENOMEM; diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a5b09e75e787..f7d2b373da0a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -541,7 +541,8 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments == 0) return -EINVAL; /* now codec parameters */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 40013b26f671..6c0b30391ba9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2177,16 +2177,11 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED) { - if (size >= runtime->start_threshold) { - err = snd_pcm_start(substream); - if (err < 0) - goto _end_unlock; - } else { - /* nothing to do */ - err = 0; + runtime->status->state == SNDRV_PCM_STATE_PREPARED && + size >= runtime->start_threshold) { + err = snd_pcm_start(substream); + if (err < 0) goto _end_unlock; - } } avail = snd_pcm_avail(substream); diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 598d140bb7cb..5fc497c6d738 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -903,6 +903,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; + if (!ins) + return 0; + snd_info_free_entry(ins->proc_sym_info_entry); ins->proc_sym_info_entry = NULL; diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 9174f1b3a987..1ec706ced75c 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -115,7 +115,8 @@ static int hda_codec_driver_probe(struct device *dev) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error_module; - if (codec->card->registered) { + /* only register after the bus probe finished; otherwise it's racy */ + if (!codec->bus->bus_probing && codec->card->registered) { err = snd_card_register(codec->card); if (err < 0) goto error_module; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e784130ea4e0..e5c49003e75f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2185,6 +2185,7 @@ static int azx_probe_continue(struct azx *chip) int dev = chip->dev_index; int err; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; /* bind with i915 if needed */ @@ -2269,6 +2270,7 @@ out_free: if (err < 0) hda->init_failed = 1; complete_all(&hda->probe_wait); + to_hda_bus(bus)->bus_probing = 0; return err; } diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e5bdbc245682..29882bda7632 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -8451,8 +8451,10 @@ static void ca0132_free(struct hda_codec *codec) ca0132_exit_chip(codec); snd_hda_power_down(codec); - if (IS_ENABLED(CONFIG_PCI) && spec->mem_base) +#ifdef CONFIG_PCI + if (spec->mem_base) pci_iounmap(codec->bus->pci, spec->mem_base); +#endif kfree(spec->spec_init_verbs); kfree(codec->spec); } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 51cc6589443f..a4ee7656d9ee 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -924,6 +924,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), @@ -931,6 +932,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aee4cbd29d53..6df758adff84 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -117,6 +117,7 @@ struct alc_spec { int codec_variant; /* flag for other variants */ unsigned int has_alc5505_dsp:1; unsigned int no_depop_delay:1; + unsigned int done_hp_init:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -514,6 +515,15 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) } } +/* get a primary headphone pin if available */ +static hda_nid_t alc_get_hp_pin(struct alc_spec *spec) +{ + if (spec->gen.autocfg.hp_pins[0]) + return spec->gen.autocfg.hp_pins[0]; + if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) + return spec->gen.autocfg.line_out_pins[0]; + return 0; +} /* * Realtek SSID verification @@ -724,9 +734,7 @@ do_sku: * 15 : 1 --> enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ - if (!spec->gen.autocfg.hp_pins[0] && - !(spec->gen.autocfg.line_out_pins[0] && - spec->gen.autocfg.line_out_type == AUTO_PIN_HP_OUT)) { + if (!alc_get_hp_pin(spec)) { hda_nid_t nid; tmp = (ass >> 11) & 0x3; /* HP to chassis */ nid = ports[tmp]; @@ -2958,7 +2966,7 @@ static void alc282_restore_default_value(struct hda_codec *codec) static void alc282_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; int coef78; @@ -2995,7 +3003,7 @@ static void alc282_init(struct hda_codec *codec) static void alc282_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; int coef78; @@ -3073,14 +3081,9 @@ static void alc283_restore_default_value(struct hda_codec *codec) static void alc283_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - alc283_restore_default_value(codec); if (!hp_pin) @@ -3114,14 +3117,9 @@ static void alc283_init(struct hda_codec *codec) static void alc283_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) { alc269_shutup(codec); return; @@ -3155,7 +3153,7 @@ static void alc283_shutup(struct hda_codec *codec) static void alc256_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) @@ -3191,7 +3189,7 @@ static void alc256_init(struct hda_codec *codec) static void alc256_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) { @@ -3227,7 +3225,7 @@ static void alc256_shutup(struct hda_codec *codec) static void alc225_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; if (!hp_pin) @@ -3270,7 +3268,7 @@ static void alc225_init(struct hda_codec *codec) static void alc225_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; if (!hp_pin) { @@ -3314,7 +3312,7 @@ static void alc225_shutup(struct hda_codec *codec) static void alc_default_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) @@ -3343,7 +3341,7 @@ static void alc_default_init(struct hda_codec *codec) static void alc_default_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) { @@ -3372,6 +3370,48 @@ static void alc_default_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc294_hp_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = alc_get_hp_pin(spec); + int i, val; + + if (!hp_pin) + return; + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + msleep(100); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ + alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ + + /* Wait for depop procedure finish */ + val = alc_read_coefex_idx(codec, 0x58, 0x01); + for (i = 0; i < 20 && val & 0x0080; i++) { + msleep(50); + val = alc_read_coefex_idx(codec, 0x58, 0x01); + } + /* Set HP depop to auto mode */ + alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b); + msleep(50); +} + +static void alc294_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->done_hp_init) { + alc294_hp_init(codec); + spec->done_hp_init = true; + } + alc_default_init(codec); +} + static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, unsigned int val) { @@ -4102,6 +4142,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: + alc_process_coef_fw(codec, alc225_pre_hsmode); alc_process_coef_fw(codec, coef0225); break; case 0x10ec0867: @@ -4736,7 +4777,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); int new_headset_mode; @@ -5015,7 +5056,7 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int hp_pin = spec->gen.autocfg.hp_pins[0]; + int hp_pin = alc_get_hp_pin(spec); /* Prevent pop noises when headphones are plugged in */ snd_hda_codec_write(codec, hp_pin, 0, @@ -5108,7 +5149,7 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PROBE) { int mic_pin = find_ext_mic_pin(codec); - int hp_pin = spec->gen.autocfg.hp_pins[0]; + int hp_pin = alc_get_hp_pin(spec); if (snd_BUG_ON(!mic_pin || !hp_pin)) return; @@ -5440,6 +5481,13 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, } } +static void alc_fixup_disable_mic_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5549,6 +5597,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC280_FIXUP_HP_HEADSET_MIC, @@ -5582,6 +5631,7 @@ enum { ALC294_FIXUP_ASUS_HEADSET_MIC, ALC294_FIXUP_ASUS_SPK, ALC225_FIXUP_HEADSET_JACK, + ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6268,6 +6318,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DISABLE_MIC_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_mic_vref, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -6277,7 +6333,7 @@ static const struct hda_fixup alc269_fixups[] = { {} }, .chained = true, - .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + .chain_id = ALC225_FIXUP_DISABLE_MIC_VREF }, [ALC280_FIXUP_HP_HEADSET_MIC] = { .type = HDA_FIXUP_FUNC, @@ -6522,6 +6578,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, }, + [ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6584,6 +6649,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0872, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0873, "Dell Precision 3930", ALC255_FIXUP_DUMMY_LINEOUT_VERB), + SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -6699,6 +6765,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), @@ -6910,7 +6977,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"}, {.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"}, {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, - {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"}, + {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, @@ -7357,37 +7424,6 @@ static void alc269_fill_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4, 0, 1<<11); } -static void alc294_hp_init(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; - int i, val; - - if (!hp_pin) - return; - - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - - msleep(100); - - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); - - alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ - alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ - - /* Wait for depop procedure finish */ - val = alc_read_coefex_idx(codec, 0x58, 0x01); - for (i = 0; i < 20 && val & 0x0080; i++) { - msleep(50); - val = alc_read_coefex_idx(codec, 0x58, 0x01); - } - /* Set HP depop to auto mode */ - alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b); - msleep(50); -} - /* */ static int patch_alc269(struct hda_codec *codec) @@ -7513,7 +7549,7 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC294; spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */ - alc294_hp_init(codec); + spec->init_hook = alc294_init; break; case 0x10ec0300: spec->codec_variant = ALC269_TYPE_ALC300; @@ -7525,7 +7561,7 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ - alc294_hp_init(codec); + spec->init_hook = alc294_init; break; } diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 022a8912c8a2..3d58338fa3cf 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -611,14 +611,16 @@ static int acp3x_audio_probe(struct platform_device *pdev) } irqflags = *((unsigned int *)(pdev->dev.platform_data)); - adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), - GFP_KERNEL); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); return -ENODEV; } + adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); + if (!adata) + return -ENOMEM; + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3ab2949c1dfa..b19d7a3e7a2c 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1890,51 +1890,31 @@ static void hdmi_codec_remove(struct snd_soc_component *component) pm_runtime_disable(&hdev->dev); } -#ifdef CONFIG_PM -static int hdmi_codec_prepare(struct device *dev) -{ - struct hdac_device *hdev = dev_to_hdac_dev(dev); - - pm_runtime_get_sync(&hdev->dev); - - /* - * Power down afg. - * codec_read is preferred over codec_write to set the power state. - * This way verb is send to set the power state and response - * is received. So setting power state is ensured without using loop - * to read the state. - */ - snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - - return 0; -} - -static void hdmi_codec_complete(struct device *dev) +#ifdef CONFIG_PM_SLEEP +static int hdmi_codec_resume(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + int ret; - /* Power up afg */ - snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - - hdac_hdmi_skl_enable_all_pins(hdev); - hdac_hdmi_skl_enable_dp12(hdev); - + ret = pm_runtime_force_resume(dev); + if (ret < 0) + return ret; /* * As the ELD notify callback request is not entertained while the * device is in suspend state. Need to manually check detection of * all pins here. pin capablity change is not support, so use the * already set pin caps. + * + * NOTE: this is safe to call even if the codec doesn't actually resume. + * The pin check involves only with DRM audio component hooks, so it + * works even if the HD-audio side is still dreaming peacefully. */ hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - - pm_runtime_put_sync(&hdev->dev); + return 0; } #else -#define hdmi_codec_prepare NULL -#define hdmi_codec_complete NULL +#define hdmi_codec_resume NULL #endif static const struct snd_soc_component_driver hdmi_hda_codec = { @@ -2135,75 +2115,6 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev) } #ifdef CONFIG_PM -/* - * Power management sequences - * ========================== - * - * The following explains the PM handling of HDAC HDMI with its parent - * device SKL and display power usage - * - * Probe - * ----- - * In SKL probe, - * 1. skl_probe_work() powers up the display (refcount++ -> 1) - * 2. enumerates the codecs on the link - * 3. powers down the display (refcount-- -> 0) - * - * In HDAC HDMI probe, - * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) - * 2. probe the codec - * 3. put the HDAC HDMI device to runtime suspend - * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * - * Once children are runtime suspended, SKL device also goes to runtime - * suspend - * - * HDMI Playback - * ------------- - * Open HDMI device, - * 1. skl_runtime_resume() invoked - * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) - * - * Close HDMI device, - * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * 2. skl_runtime_suspend() invoked - * - * S0/S3 Cycle with playback in progress - * ------------------------------------- - * When the device is opened for playback, the device is runtime active - * already and the display refcount is 1 as explained above. - * - * Entering to S3, - * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just - * increments the PM runtime usage count of the codec since the device - * is in use already - * 2. skl_suspend() powers down the display (refcount-- -> 0) - * - * Wakeup from S3, - * 1. skl_resume() powers up the display (refcount++ -> 1) - * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just - * decrements the PM runtime usage count of the codec since the device - * is in use already - * - * Once playback is stopped, the display refcount is set to 0 as explained - * above in the HDMI playback sequence. The PM handlings are designed in - * such way that to balance the refcount of display power when the codec - * device put to S3 while playback is going on. - * - * S0/S3 Cycle without playback in progress - * ---------------------------------------- - * Entering to S3, - * 1. hdmi_codec_prepare() invoke the runtime resume of codec - * 2. skl_runtime_resume() invoked - * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) - * 4. skl_suspend() powers down the display (refcount-- -> 0) - * - * Wakeup from S3, - * 1. skl_resume() powers up the display (refcount++ -> 1) - * 2. hdmi_codec_complete() invokes the runtime suspend of codec - * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * 4. skl_runtime_suspend() invoked - */ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); @@ -2277,8 +2188,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev) static const struct dev_pm_ops hdac_hdmi_pm = { SET_RUNTIME_PM_OPS(hdac_hdmi_runtime_suspend, hdac_hdmi_runtime_resume, NULL) - .prepare = hdmi_codec_prepare, - .complete = hdmi_codec_complete, + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, hdmi_codec_resume) }; static const struct hda_device_id hdmi_list[] = { diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d00734d31e04..e5b6769b9797 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -795,6 +795,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) if (hcd->spdif) hcp->daidrv[i] = hdmi_spdif_dai; + dev_set_drvdata(dev, hcp); + ret = devm_snd_soc_register_component(dev, &hdmi_driver, hcp->daidrv, dai_count); if (ret) { @@ -802,8 +804,6 @@ static int hdmi_codec_probe(struct platform_device *pdev) __func__, ret); return ret; } - - dev_set_drvdata(dev, hcp); return 0; } diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 6cb1653be804..4cc24a5d5c31 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1400,24 +1400,20 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) if (ret != 0) { dev_err(component->dev, "Failed to set digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, PCM512x_ANALOG_MUTE_DET, mute_det, (mute_det & 0x3) == 0, 200, 10000); - - mutex_unlock(&pcm512x->mutex); } else { pcm512x->mute &= ~0x1; ret = pcm512x_update_mute(pcm512x); if (ret != 0) { dev_err(component->dev, "Failed to update digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, @@ -1428,9 +1424,10 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) 200, 10000); } +unlock: mutex_unlock(&pcm512x->mutex); - return 0; + return ret; } static const struct snd_soc_dai_ops pcm512x_dai_ops = { diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 0ef966d56bac..e2855ab9a2c6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 4d46f4567c3a..bec2eefa8b0f 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -280,6 +280,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) rt5514_dsp = devm_kzalloc(component->dev, sizeof(*rt5514_dsp), GFP_KERNEL); + if (!rt5514_dsp) + return -ENOMEM; rt5514_dsp->dev = &rt5514_spi->dev; mutex_init(&rt5514_dsp->dma_lock); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 34cfaf8f6f34..a9b91bcfcc09 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1778,7 +1778,9 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "DA ASRC"}, {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "AD ASRC"}, {"DAC STO1 ASRC", NULL, "DA ASRC"}, {"DAC STO1 ASRC", NULL, "CLKDET"}, @@ -2512,6 +2514,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); mutex_unlock(&rt5682->calibrate_mutex); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d82a8301fd74..96944cff0ed7 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -849,18 +849,18 @@ #define RT5682_SCLK_SRC_PLL2 (0x2 << 13) #define RT5682_SCLK_SRC_SDW (0x3 << 13) #define RT5682_SCLK_SRC_RCCLK (0x4 << 13) -#define RT5682_PLL1_SRC_MASK (0x3 << 10) -#define RT5682_PLL1_SRC_SFT 10 -#define RT5682_PLL1_SRC_MCLK (0x0 << 10) -#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) -#define RT5682_PLL1_SRC_SDW (0x2 << 10) -#define RT5682_PLL1_SRC_RC (0x3 << 10) -#define RT5682_PLL2_SRC_MASK (0x3 << 8) -#define RT5682_PLL2_SRC_SFT 8 -#define RT5682_PLL2_SRC_MCLK (0x0 << 8) -#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) -#define RT5682_PLL2_SRC_SDW (0x2 << 8) -#define RT5682_PLL2_SRC_RC (0x3 << 8) +#define RT5682_PLL2_SRC_MASK (0x3 << 10) +#define RT5682_PLL2_SRC_SFT 10 +#define RT5682_PLL2_SRC_MCLK (0x0 << 10) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL2_SRC_SDW (0x2 << 10) +#define RT5682_PLL2_SRC_RC (0x3 << 10) +#define RT5682_PLL1_SRC_MASK (0x3 << 8) +#define RT5682_PLL1_SRC_SFT 8 +#define RT5682_PLL1_SRC_MCLK (0x0 << 8) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL1_SRC_SDW (0x2 << 8) +#define RT5682_PLL1_SRC_RC (0x3 << 8) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e2b5a11b16d1..f03195d2ab2e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -822,6 +822,10 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + /* Initial cold start */ + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + break; + /* Switch off BCLK_N Divider */ snd_soc_component_update_bits(component, AIC32X4_BCLKN, AIC32X4_BCLKEN, 0); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 392d5eef356d..99e07b01a2ce 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk input"); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk input"); } - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\nData received from %s\n", audmux_port_string((pdcr >> 13) & 0x7)); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 99a62ba409df..bd9fd2035c55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index afc559866095..91a2436ce952 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + int ret; + + ret = + snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + if (ret) + return ret; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); return 0; } diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 68e6543e6cb0..99f2a0156ae8 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c74c4f17316f..8f83b182c4f9 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -55,39 +55,6 @@ enum { GLK_DPCM_AUDIO_HDMI3_PB, }; -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret = 0; - - codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop sysclk: %d\n", ret); - } else if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); - if (ret < 0) { - dev_err(card->dev, "can't set codec pll: %d\n", ret); - return ret; - } - } - - if (ret) - dev_err(card->dev, "failed to start internal clk: %d\n", ret); - - return ret; -} - static const struct snd_kcontrol_new geminilake_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -102,14 +69,10 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route geminilake_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -117,7 +80,6 @@ static const struct snd_soc_dapm_route geminilake_map[] = { { "Spk", NULL, "Speaker" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, /* digital mics */ @@ -177,6 +139,13 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_jack *jack; int ret; + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + /* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index eab1f439dd3f..a4022983a7ce 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 60c94836bf5b..4ed5b7e17d44 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -336,9 +336,6 @@ static int skl_suspend(struct device *dev) skl->skl_sst->fw_loaded = false; } - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); - return 0; } @@ -350,10 +347,6 @@ static int skl_resume(struct device *dev) struct hdac_ext_link *hlink = NULL; int ret; - /* Turned OFF in HDMI codec driver after codec reconfiguration */ - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true); - /* * resume only when we are not in suspend active, otherwise need to * restore the device @@ -446,8 +439,10 @@ static int skl_free(struct hdac_bus *bus) snd_hdac_ext_bus_exit(bus); cancel_work_sync(&skl->probe_work); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); snd_hdac_i915_exit(bus); + } return 0; } @@ -814,7 +809,7 @@ static void skl_probe_work(struct work_struct *work) err = skl_platform_register(bus->dev); if (err < 0) { dev_err(bus->dev, "platform register failed: %d\n", err); - return; + goto out_err; } err = skl_machine_device_register(skl); diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 5b986b74dd36..548eb4fa2da6 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -570,10 +570,10 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)compress_event_handler, prtd, stream_id, LEGACY_PCM_MODE); - if (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { dev_err(dev, "Could not allocate memory\n"); - kfree(prtd); - return -ENOMEM; + ret = PTR_ERR(prtd->audio_client); + goto free_prtd; } size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * @@ -582,7 +582,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) &prtd->dma_buffer); if (ret) { dev_err(dev, "Cannot allocate buffer(s)\n"); - return ret; + goto free_client; } if (pdata->sid < 0) @@ -595,6 +595,13 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) runtime->private_data = prtd; return 0; + +free_client: + q6asm_audio_client_free(prtd->audio_client); +free_prtd: + kfree(prtd); + + return ret; } static int q6asm_dai_compr_free(struct snd_compr_stream *stream) @@ -874,7 +881,7 @@ static int of_q6asm_parse_dai_data(struct device *dev, for_each_child_of_node(dev->of_node, node) { ret = of_property_read_u32(node, "reg", &id); - if (ret || id > MAX_SESSIONS || id < 0) { + if (ret || id >= MAX_SESSIONS || id < 0) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 1db8ef668223..6f66a58e23ca 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -158,17 +158,24 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static void sdm845_jack_free(struct snd_jack *jack) +{ + struct snd_soc_component *component = jack->private_data; + + snd_soc_component_set_jack(component, NULL, NULL); +} + static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); - int i, rval; + struct snd_jack *jack; + int rval; if (!pdata->jack_setup) { - struct snd_jack *jack; - rval = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_HEADPHONE | @@ -190,16 +197,22 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + jack = pdata->jack.jack; + component = codec_dai->component; - component = dai->component; - rval = snd_soc_component_set_jack( - component, &pdata->jack, NULL); + jack->private_data = component; + jack->private_free = sdm845_jack_free; + rval = snd_soc_component_set_jack(component, + &pdata->jack, NULL); if (rval != 0 && rval != -ENOTSUPP) { dev_warn(card->dev, "Failed to set jack: %d\n", rval); return rval; } + break; + default: + break; } return 0; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d6c62aa13041..ce00fe2f6aae 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -700,6 +700,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, { struct i2s_dai *i2s = to_info(dai); u32 mod, mask = 0, val = 0; + struct clk *rclksrc; unsigned long flags; WARN_ON(!pm_runtime_active(dai->dev)); @@ -782,6 +783,10 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, i2s->frmclk = params_rate(params); + rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; + if (rclksrc && !IS_ERR(rclksrc)) + i2s->rclk_srcrate = clk_get_rate(rclksrc); + return 0; } @@ -886,11 +891,6 @@ static int config_setup(struct i2s_dai *i2s) return 0; if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { - struct clk *rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; - - if (rclksrc && !IS_ERR(rclksrc)) - i2s->rclk_srcrate = clk_get_rate(rclksrc); - psr = i2s->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); dev_dbg(&i2s->pdev->dev, diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 59e250cc2e9d..e819e965e1db 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1526,14 +1526,14 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int ret; /* - * 1) Avoid duplicate register (ex. MIXer case) - * 2) re-register if card was rebinded + * 1) Avoid duplicate register for DVC with MIX case + * 2) Allow duplicate register for MIX + * 3) re-register if card was rebinded */ list_for_each_entry(kctrl, &card->controls, list) { struct rsnd_kctrl_cfg *c = kctrl->private_data; - if (strcmp(kctrl->id.name, name) == 0 && - c->mod == mod) + if (c == cfg) return 0; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 45ef295743ec..f5afab631abb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->usrcnt > 1) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index c5934adcfd01..c74991dd18ab 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -79,7 +79,7 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, break; case 9: for (i = 0; i < 4; i++) - rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << (id * 4)); + rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << 4); break; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..50617db05c46 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -735,14 +735,19 @@ static struct snd_soc_component *soc_find_component( const struct device_node *of_node, const char *name) { struct snd_soc_component *component; + struct device_node *component_of_node; lockdep_assert_held(&client_mutex); for_each_component(component) { if (of_node) { - if (component->dev->of_node == of_node) + component_of_node = component->dev->of_node; + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + if (component_of_node == of_node) return component; - } else if (strcmp(component->name, name) == 0) { + } else if (name && strcmp(component->name, name) == 0) { return component; } } @@ -951,7 +956,7 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (!dai || !dai->probed || + if (!dai || !dai->probed || !dai->driver || dai->driver->remove_order != order) return; @@ -1034,17 +1039,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { + if (!platform || dai_link->legacy_platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); if (!platform) return -ENOMEM; - dai_link->platform = platform; - platform->name = dai_link->platform_name; - platform->of_node = dai_link->platform_of_node; - platform->dai_name = NULL; + dai_link->platform = platform; + dai_link->legacy_platform = 1; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; } /* if there's no platform we match on the empty platform */ @@ -1129,6 +1135,15 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if ((link->platform->of_node || link->platform->name) && + !soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1155,15 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if ((link->cpu_of_node || link->cpu_name) && + !soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified @@ -2739,15 +2763,18 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + mutex_lock(&client_mutex); for_each_card_prelinks(card, i, link) { ret = soc_init_dai_link(card, link); if (ret) { dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); + mutex_unlock(&client_mutex); return ret; } } + mutex_unlock(&client_mutex); dev_set_drvdata(card->dev, card); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..20bad755888b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -83,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -100,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, @@ -2019,19 +2043,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w, NULL, NULL); } - ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", w->force ? " (forced)" : "", in, out); if (w->reg >= 0) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " - R%d(0x%x) mask 0x%x", w->reg, w->reg, w->mask << w->shift); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (w->sname) - ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, w->active ? "active" : "inactive"); @@ -2044,7 +2068,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " %s \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 045ef136903d..fc79ec6927e3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp, { struct snd_soc_dai_driver *dai_drv = container_of(dobj, struct snd_soc_dai_driver, dobj); + struct snd_soc_dai *dai; if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp, if (dobj->ops && dobj->ops->dai_unload) dobj->ops->dai_unload(comp, dobj); + list_for_each_entry(dai, &comp->dai_list, list) + if (dai->driver == dai_drv) + dai->driver = NULL; + kfree(dai_drv->name); list_del(&dobj->list); kfree(dai_drv); diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index eeda6d5565bc..a10fcb5963c6 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -108,7 +108,7 @@ struct davinci_mcasp { /* Used for comstraint setting on the second stream */ u32 channels; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM struct davinci_mcasp_context context; #endif @@ -1486,74 +1486,6 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) return 0; } -#ifdef CONFIG_PM_SLEEP -static int davinci_mcasp_suspend(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - context->pm_state = pm_runtime_active(mcasp->dev); - if (!context->pm_state) - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); - } - - for (i = 0; i < mcasp->num_serializer; i++) - context->xrsr_regs[i] = mcasp_get_reg(mcasp, - DAVINCI_MCASP_XRSRCTL_REG(i)); - - pm_runtime_put_sync(mcasp->dev); - - return 0; -} - -static int davinci_mcasp_resume(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); - } - - for (i = 0; i < mcasp->num_serializer; i++) - mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), - context->xrsr_regs[i]); - - if (!context->pm_state) - pm_runtime_put_sync(mcasp->dev); - - return 0; -} -#else -#define davinci_mcasp_suspend NULL -#define davinci_mcasp_resume NULL -#endif - #define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ @@ -1571,8 +1503,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", .probe = davinci_mcasp_dai_probe, - .suspend = davinci_mcasp_suspend, - .resume = davinci_mcasp_resume, .playback = { .channels_min = 1, .channels_max = 32 * 16, @@ -1976,7 +1906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } mcasp->num_serializer = pdata->num_serializer; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev, mcasp->num_serializer, sizeof(u32), GFP_KERNEL); @@ -2196,11 +2126,73 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int davinci_mcasp_runtime_suspend(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); + } + + for (i = 0; i < mcasp->num_serializer; i++) + context->xrsr_regs[i] = mcasp_get_reg(mcasp, + DAVINCI_MCASP_XRSRCTL_REG(i)); + + return 0; +} + +static int davinci_mcasp_runtime_resume(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); + } + + for (i = 0; i < mcasp->num_serializer; i++) + mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + context->xrsr_regs[i]); + + return 0; +} + +#endif + +static const struct dev_pm_ops davinci_mcasp_pm_ops = { + SET_RUNTIME_PM_OPS(davinci_mcasp_runtime_suspend, + davinci_mcasp_runtime_resume, + NULL) +}; + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 25e287feb58c..723a583a8d57 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -1,5 +1,5 @@ config SND_SOC_XILINX_I2S - tristate "Audio support for the the Xilinx I2S" + tristate "Audio support for the Xilinx I2S" help Select this option to enable Xilinx I2S Audio. This enables I2S playback and capture using xilinx soft IP. In transmitter diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c index d4ae9eff41ce..8b353166ad44 100644 --- a/sound/soc/xilinx/xlnx_i2s.c +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -1,12 +1,11 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Xilinx ASoC I2S audio support - * - * Copyright (C) 2018 Xilinx, Inc. - * - * Author: Praveen Vuppala <praveenv@xilinx.com> - * Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com> - */ +// +// Xilinx ASoC I2S audio support +// +// Copyright (C) 2018 Xilinx, Inc. +// +// Author: Praveen Vuppala <praveenv@xilinx.com> +// Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com> #include <linux/io.h> #include <linux/module.h> diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 7609eceba1a2..9e71d7cda999 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2541,8 +2541,8 @@ static int snd_dbri_create(struct snd_card *card, dbri->op = op; dbri->irq = irq; - dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma), - &dbri->dma_dvma, GFP_KERNEL); + dbri->dma = dma_alloc_coherent(&op->dev, sizeof(struct dbri_dma), + &dbri->dma_dvma, GFP_KERNEL); if (!dbri->dma) return -ENOMEM; diff --git a/sound/usb/card.c b/sound/usb/card.c index a105947eaf55..746a72e23cf9 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -246,7 +246,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) h1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, UAC_HEADER); - if (!h1) { + if (!h1 || h1->bLength < sizeof(*h1)) { dev_err(&dev->dev, "cannot find UAC_HEADER\n"); return -EINVAL; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c63c84b54969..e7d441d0e839 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -753,8 +753,9 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { int mu_channels; + void *c; - if (desc->bLength < 11) + if (desc->bLength < sizeof(*desc)) return -EINVAL; if (!desc->bNrInPins) return -EINVAL; @@ -763,6 +764,8 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, case UAC_VERSION_1: case UAC_VERSION_2: default: + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 1) + return 0; /* no bmControls -> skip */ mu_channels = uac_mixer_unit_bNrChannels(desc); break; case UAC_VERSION_3: @@ -772,7 +775,11 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, } if (!mu_channels) - return -EINVAL; + return 0; + + c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); + if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength) + return 0; /* no bmControls -> skip */ return mu_channels; } @@ -944,7 +951,7 @@ static int check_input_term(struct mixer_build *state, int id, struct uac_mixer_unit_descriptor *d = p1; err = uac_mixer_unit_get_channels(state, d); - if (err < 0) + if (err <= 0) return err; term->channels = err; @@ -2068,11 +2075,15 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, if (state->mixer->protocol == UAC_VERSION_2) { struct uac2_input_terminal_descriptor *d_v2 = raw_desc; + if (d_v2->bLength < sizeof(*d_v2)) + return -EINVAL; control = UAC2_TE_CONNECTOR; term_id = d_v2->bTerminalID; bmctls = le16_to_cpu(d_v2->bmControls); } else if (state->mixer->protocol == UAC_VERSION_3) { struct uac3_input_terminal_descriptor *d_v3 = raw_desc; + if (d_v3->bLength < sizeof(*d_v3)) + return -EINVAL; control = UAC3_TE_INSERTION; term_id = d_v3->bTerminalID; bmctls = le32_to_cpu(d_v3->bmControls); @@ -2118,7 +2129,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, if (err < 0) continue; /* no bmControls field (e.g. Maya44) -> ignore */ - if (desc->bLength <= 10 + input_pins) + if (!num_outs) continue; err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0) @@ -2314,7 +2325,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, char *name) { struct uac_processing_unit_descriptor *desc = raw_desc; - int num_ins = desc->bNrInPins; + int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; int i, err, nameid, type, len; @@ -2329,7 +2340,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, 0, NULL, default_value_info }; - if (desc->bLength < 13 || desc->bLength < 13 + num_ins || + if (desc->bLength < 13) { + usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); + return -EINVAL; + } + + num_ins = desc->bNrInPins; + if (desc->bLength < 13 + num_ins || desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) { usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); return -EINVAL; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 382847154227..db114f3977e0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -314,6 +314,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } +/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk + * applies. Returns 1 if a quirk was found. + */ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_device *dev, struct usb_interface_descriptor *altsd, @@ -384,7 +387,7 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; - return 0; + return 1; } static int set_sync_endpoint(struct snd_usb_substream *subs, @@ -423,6 +426,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, if (err < 0) return err; + /* endpoint set by quirk */ + if (err > 0) + return 0; + if (altsd->bNumEndpoints < 2) return 0; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 37fc0447c071..b345beb447bd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3326,6 +3326,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, + { + .ifnum = -1 + }, } } }, @@ -3369,6 +3372,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, + { + .ifnum = -1 + }, } } }, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 96340f23f86d..7e65fe853ee3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -768,7 +768,7 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) * REG1: PLL binary search enable, soft mute enable. */ CM6206_REG1_PLLBIN_EN | - CM6206_REG1_SOFT_MUTE_EN | + CM6206_REG1_SOFT_MUTE_EN, /* * REG2: enable output drivers, * select front channels to the headphone output, @@ -1492,6 +1492,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ @@ -1566,6 +1567,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x20b1: /* XMOS based devices */ case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ + case 0x2ab6: /* T+A devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 67cf849aa16b..d9e3de495c16 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -596,12 +596,8 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, csep = snd_usb_find_desc(alts->extra, alts->extralen, NULL, USB_DT_CS_ENDPOINT); if (!csep || csep->bLength < 7 || - csep->bDescriptorSubtype != UAC_EP_GENERAL) { - usb_audio_warn(chip, - "%u:%d : no or invalid class specific endpoint descriptor\n", - iface_no, altsd->bAlternateSetting); - return 0; - } + csep->bDescriptorSubtype != UAC_EP_GENERAL) + goto error; if (protocol == UAC_VERSION_1) { attributes = csep->bmAttributes; @@ -609,6 +605,8 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct uac2_iso_endpoint_descriptor *csep2 = (struct uac2_iso_endpoint_descriptor *) csep; + if (csep2->bLength < sizeof(*csep2)) + goto error; attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX; /* emulate the endpoint attributes of a v1 device */ @@ -618,12 +616,20 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct uac3_iso_endpoint_descriptor *csep3 = (struct uac3_iso_endpoint_descriptor *) csep; + if (csep3->bLength < sizeof(*csep3)) + goto error; /* emulate the endpoint attributes of a v1 device */ if (le32_to_cpu(csep3->bmControls) & UAC2_CONTROL_PITCH) attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL; } return attributes; + + error: + usb_audio_warn(chip, + "%u:%d : no or invalid class specific endpoint descriptor\n", + iface_no, altsd->bAlternateSetting); + return 0; } /* find an input terminal descriptor (either UAC1 or UAC2) with the given @@ -631,13 +637,17 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, */ static void * snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) + int terminal_id, bool uac23) { struct uac2_input_terminal_descriptor *term = NULL; + size_t minlen = uac23 ? sizeof(struct uac2_input_terminal_descriptor) : + sizeof(struct uac_input_terminal_descriptor); while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_INPUT_TERMINAL))) { + if (term->bLength < minlen) + continue; if (term->bTerminalID == terminal_id) return term; } @@ -655,7 +665,8 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_OUTPUT_TERMINAL))) { - if (term->bTerminalID == terminal_id) + if (term->bLength >= sizeof(*term) && + term->bTerminalID == terminal_id) return term; } @@ -729,7 +740,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, format = le16_to_cpu(as->wFormatTag); /* remember the format value */ iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + false); if (iterm) { num_channels = iterm->bNrChannels; chconfig = le16_to_cpu(iterm->wChannelConfig); @@ -764,7 +776,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, * to extract the clock */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + true); if (input_term) { clock = input_term->bCSourceID; if (!chconfig && (num_channels == input_term->bNrChannels)) @@ -998,7 +1011,8 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * to extract the clock */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + true); if (input_term) { clock = input_term->bCSourceID; goto found_clock; |