diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt index 34cf70e2cbc4720510cf03e05eba6f3bcb6b2cae..4bb513ae62fc67797857ddf11f433d49343b7743 100644 --- a/Documentation/devicetree/bindings/sound/tdm-slot.txt +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -14,8 +14,8 @@ For instance: dai-tdm-slot-tx-mask = <0 1>; dai-tdm-slot-rx-mask = <1 0>; -And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() -to specify a explicit mapping of the channels and the slots. If it's absent +And for each specified driver, there could be one .of_xlate_tdm_slot_mask() +to specify an explicit mapping of the channels and the slots. If it's absent the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the tx and rx masks. diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 47838f57a64779e3c99780eb99f39a6914d23c45..9630d252394805bb5c223f0f080a86164a33e101 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -138,6 +138,16 @@ void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq); +static void snd_pcm_stream_lock_nested(struct snd_pcm_substream *substream) +{ + struct snd_pcm_group *group = &substream->self_group; + + if (substream->pcm->nonatomic) + mutex_lock_nested(&group->mutex, SINGLE_DEPTH_NESTING); + else + spin_lock_nested(&group->lock, SINGLE_DEPTH_NESTING); +} + /** * snd_pcm_stream_unlock_irq - Unlock the PCM stream * @substream: PCM substream @@ -2166,6 +2176,12 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) } pcm_file = f.file->private_data; substream1 = pcm_file->substream; + + if (substream == substream1) { + res = -EINVAL; + goto _badf; + } + group = kzalloc(sizeof(*group), GFP_KERNEL); if (!group) { res = -ENOMEM; @@ -2194,7 +2210,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) snd_pcm_stream_unlock_irq(substream); snd_pcm_group_lock_irq(target_group, nonatomic); - snd_pcm_stream_lock(substream1); + snd_pcm_stream_lock_nested(substream1); snd_pcm_group_assign(substream1, target_group); refcount_inc(&target_group->refs); snd_pcm_stream_unlock(substream1); @@ -2210,7 +2226,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) static void relink_to_local(struct snd_pcm_substream *substream) { - snd_pcm_stream_lock(substream); + snd_pcm_stream_lock_nested(substream); snd_pcm_group_assign(substream, &substream->self_group); snd_pcm_stream_unlock(substream); } diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index ddb7c2ce3f7c909de568ade69bd7a3d95becd6b9..def8161cde4c22f35e12d3f09b608de82b618ef2 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,7 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2) + if (reg < 0x49 && channel_id <= 2) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0aa778ff7f2b2cf7ffbaed5cdafbe9a2526bd2ef..6d73f8beadb6e17f84dbeeb16e54a2df413b0fb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8161,6 +8161,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, {0x17, 0x90170110}), + SND_HDA_PIN_QUIRK(0x10ec0623, 0x17aa, "Lenovo", ALC283_FIXUP_HEADSET_MIC, + {0x14, 0x01014010}, + {0x17, 0x90170120}, + {0x18, 0x02a11030}, + {0x19, 0x02a1103f}, + {0x21, 0x0221101f}), {} }; diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b9ce44dda886f7113196883e88d50e64e3de5565..0d63ebfbff2f999160f95d1e471d6787e62ca04a 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -754,6 +754,7 @@ static struct snd_soc_dai_driver max98390_dai[] = { static int max98390_dsm_init(struct snd_soc_component *component) { int ret; + int param_size, param_start_addr; char filename[128]; const char *vendor, *product; struct max98390_priv *max98390 = @@ -778,16 +779,31 @@ static int max98390_dsm_init(struct snd_soc_component *component) } dev_dbg(component->dev, - "max98390: param fw size %ld\n", + "max98390: param fw size %zd\n", fw->size); + if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } dsm_param = (char *)fw->data; + param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; + param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; + if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || + param_start_addr < DSM_STBASS_HPF_B0_BYTE0 || + fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } + regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; - regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0, - dsm_param, - fw->size - MAX98390_DSM_PAYLOAD_OFFSET); - release_firmware(fw); + regmap_bulk_write(max98390->regmap, param_start_addr, + dsm_param, param_size); regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01); +err_alloc: + release_firmware(fw); err: return ret; } diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index f59cb114d95798c433f907d1f9d29a7f39bf4898..5f444e7779b0a38ae54389b563f3159b6ca284d9 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,8 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PAYLOAD_OFFSET_2 495 +#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { struct regmap *regmap; diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 2586d1cafc0cda31c2f5e22cc10abe04f8c17ca5..8c9daf32bab8460f797e817bc10e567a830d0856 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -80,8 +80,8 @@ int rl6231_calc_dmic_clk(int rate) for (i = 0; i < ARRAY_SIZE(div); i++) { if ((div[i] % 3) == 0) continue; - /* find divider that gives DMIC frequency below 3.072MHz */ - if (3072000 * div[i] >= rate) + /* find divider that gives DMIC frequency below 1.536MHz */ + if (1536000 * div[i] >= rate) return i; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6ba1849a77b086860b0b445b57450c6977858347..e2e1d5b03b38114b5c8d1bb90e64fa68fb75345c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data asus_t101ha_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, + .dmic2_data_pin = RT5645_DMIC2_DISABLE, + .jd_mode = 3, +}; + static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .jd_mode = 3, .in2_diff = true, @@ -3708,6 +3714,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&asus_t100ha_platform_data, }, + { + .ident = "ASUS T101HA", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"), + }, + .driver_data = (void *)&asus_t101ha_platform_data, + }, { .ident = "MINIX Z83-4", .matches = { diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index cf4feb835743db88d1c6696f12d630e6c3ffeeb0..00be73900888449a6c8139cbd762195507dee44e 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -581,7 +581,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_err(&pdev->dev, "failed to find codec device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto asrc_fail; } diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 30f70bbdf89c729dee0cda88fb65a035b9c2207b..1fdb70b9e478865919f04a56c35c1e1a41317485 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 48eda1a8aa6cbe0d07dbb67420e1428044b34980..954ab01f695b8f8a117ca4674dcdabd480f81576 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -407,7 +407,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .name = "Glk Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index cc9b5eab8b4a5f54a15b8a2dc5b6a23074d53c2a..e29c31ffd241f1d1adf255ca2a1863d5d9dfb902 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -692,7 +692,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, @@ -858,7 +858,7 @@ static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 658a9da3a40fd1bc76c7130437a619276040f7c6..09ba55fc36d5180630ca2b99e73fe94c24cf32c1 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -672,7 +672,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3fc8445d4e619dcc4ea578a8738eb..b34cf6cf11395832c23b6ec4a3b6dc37380cfab6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -566,7 +566,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 2e9b56b29d313103d61a0a4efb4ba5414e409cae..b2e867113226b5f4e0b641b3db42caba880cdd4b 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + + return 0; +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } EXPORT_SYMBOL_GPL(axg_fifo_pcm_open); diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 2ca8c98e204f2bdf5f0d726647d814dce1f20877..5a4a91c887347f5b37574cff2f9ada7c40ba0b5b 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card, links = krealloc(priv->card.dai_link, num_links * sizeof(*priv->card.dai_link), GFP_KERNEL | __GFP_ZERO); + if (!links) + goto err_links; + ldata = krealloc(priv->link_data, num_links * sizeof(*priv->link_data), GFP_KERNEL | __GFP_ZERO); - - if (!links || !ldata) { - dev_err(priv->card.dev, "failed to allocate links\n"); - return -ENOMEM; - } + if (!ldata) + goto err_ldata; priv->card.dai_link = links; priv->link_data = ldata; priv->card.num_links = num_links; return 0; + +err_ldata: + kfree(links); +err_links: + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } EXPORT_SYMBOL_GPL(meson_card_reallocate_links); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b07eca2c6cccbd56da13a3453f1dac9fb92ad1e9..7b387202c5dbd9ac4be7a243f0a120c9bf05a2ef 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1648,9 +1648,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platforms->name = component->name; /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + if (!dai_link->no_pcm) { + dai_link->no_pcm = 1; + + if (dai_link->dpcm_playback) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n", + dai_link->name); + if (dai_link->dpcm_capture) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n", + dai_link->name); + + /* convert normal link into DPCM one */ + if (!(dai_link->dpcm_playback || + dai_link->dpcm_capture)) { + dai_link->dpcm_playback = !dai_link->capture_only; + dai_link->dpcm_capture = !dai_link->playback_only; + } + } /* * override any BE fixups diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 276505fb9d50408a3a6470b569b164ad39002ba3..2c114b4542ce4cf765bf6818975da20964ba57a0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2789,20 +2789,44 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int stream; int i; + if (rtd->dai_link->dynamic && rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU for Front-Ends yet\n"); + return -EINVAL; + } + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - cpu_dai = asoc_rtd_to_cpu(rtd, 0); - if (rtd->num_cpus > 1) { - dev_err(rtd->dev, - "DPCM doesn't support Multi CPU yet\n"); - return -EINVAL; + if (rtd->dai_link->dpcm_playback) { + stream = SNDRV_PCM_STREAM_PLAYBACK; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support playback\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + playback = 1; + } + if (rtd->dai_link->dpcm_capture) { + stream = SNDRV_PCM_STREAM_CAPTURE; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support capture\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + capture = 1; } - - playback = rtd->dai_link->dpcm_playback && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); - capture = rtd->dai_link->dpcm_capture && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index ce053ba8f2e82c1d3587ef2437fa2847d9bb3750..d03b5be31255b3afa6cc7cef0e43f2358a66f4b8 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].platforms->name = dev_name(dev); links[i].codecs->dai_name = "snd-soc-dummy-dai"; links[i].codecs->name = "snd-soc-dummy"; - links[i].dpcm_playback = 1; - links[i].dpcm_capture = 1; + if (ops->drv[i].playback.channels_min) + links[i].dpcm_playback = 1; + if (ops->drv[i].capture.channels_min) + links[i].dpcm_capture = 1; } card->dai_link = links; diff --git a/sound/usb/card.c b/sound/usb/card.c index fd6fd1726ea0bdbfddb33d40a32fff5ad863f674..162bdd6eb4d4ba98f7b7281c73ddb20d7d75af83 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -634,7 +634,6 @@ static int usb_audio_probe(struct usb_interface *intf, id, &chip); if (err < 0) goto __error; - chip->pm_intf = intf; break; } else if (vid[i] != -1 || pid[i] != -1) { dev_info(&dev->dev, @@ -651,6 +650,13 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } } + + if (chip->num_interfaces >= MAX_CARD_INTERFACES) { + dev_info(&dev->dev, "Too many interfaces assigned to the single USB-audio card\n"); + err = -EINVAL; + goto __error; + } + dev_set_drvdata(&dev->dev, chip); /* @@ -703,6 +709,7 @@ static int usb_audio_probe(struct usb_interface *intf, } usb_chip[chip->index] = chip; + chip->intf[chip->num_interfaces] = intf; chip->num_interfaces++; usb_set_intfdata(intf, chip); atomic_dec(&chip->active); @@ -818,19 +825,37 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) int snd_usb_autoresume(struct snd_usb_audio *chip) { + int i, err; + if (atomic_read(&chip->shutdown)) return -EIO; - if (atomic_inc_return(&chip->active) == 1) - return usb_autopm_get_interface(chip->pm_intf); + if (atomic_inc_return(&chip->active) != 1) + return 0; + + for (i = 0; i < chip->num_interfaces; i++) { + err = usb_autopm_get_interface(chip->intf[i]); + if (err < 0) { + /* rollback */ + while (--i >= 0) + usb_autopm_put_interface(chip->intf[i]); + atomic_dec(&chip->active); + return err; + } + } return 0; } void snd_usb_autosuspend(struct snd_usb_audio *chip) { + int i; + if (atomic_read(&chip->shutdown)) return; - if (atomic_dec_and_test(&chip->active)) - usb_autopm_put_interface(chip->pm_intf); + if (!atomic_dec_and_test(&chip->active)) + return; + + for (i = 0; i < chip->num_interfaces; i++) + usb_autopm_put_interface(chip->intf[i]); } static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) @@ -843,9 +868,6 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (chip == (void *)-1L) return 0; - chip->autosuspended = !!PMSG_IS_AUTO(message); - if (!chip->autosuspended) - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); if (!chip->num_suspended_intf++) { list_for_each_entry(as, &chip->pcm_list, list) { snd_usb_pcm_suspend(as); @@ -858,6 +880,11 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) snd_usb_mixer_suspend(mixer); } + if (!PMSG_IS_AUTO(message) && !chip->system_suspend) { + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + chip->system_suspend = chip->num_suspended_intf; + } + return 0; } @@ -871,10 +898,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) if (chip == (void *)-1L) return 0; - if (--chip->num_suspended_intf) - return 0; atomic_inc(&chip->active); /* avoid autopm */ + if (chip->num_suspended_intf > 1) + goto out; list_for_each_entry(as, &chip->pcm_list, list) { err = snd_usb_pcm_resume(as); @@ -896,9 +923,12 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) snd_usbmidi_resume(p); } - if (!chip->autosuspended) + out: + if (chip->num_suspended_intf == chip->system_suspend) { snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); - chip->autosuspended = 0; + chip->system_suspend = 0; + } + chip->num_suspended_intf--; err_out: atomic_dec(&chip->active); /* allow autopm after this point */ diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 6d6492195bdc7e7a110295f204649c4b9355f64f..4ec491011b19c2ba7196226dd2240a25eac065ce 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -40,6 +40,18 @@ .ifnum = QUIRK_NO_INTERFACE \ } +/* HP Thunderbolt Dock Audio Headset */ +{ + USB_DEVICE(0x03f0, 0x0269), + QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Headset", + "HP-Thunderbolt-Dock-Audio-Headset"), +}, +/* HP Thunderbolt Dock Audio Module */ +{ + USB_DEVICE(0x03f0, 0x0567), + QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Module", + "HP-Thunderbolt-Dock-Audio-Module"), +}, /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1c892c7f14d78b6595f1a97a4e67aaad3df4a286..b91c4c0807eca41fc22753c0cfb6ce3e60b975f8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -19,14 +19,16 @@ struct media_device; struct media_intf_devnode; +#define MAX_CARD_INTERFACES 16 + struct snd_usb_audio { int index; struct usb_device *dev; struct snd_card *card; - struct usb_interface *pm_intf; + struct usb_interface *intf[MAX_CARD_INTERFACES]; u32 usb_id; struct mutex mutex; - unsigned int autosuspended:1; + unsigned int system_suspend; atomic_t active; atomic_t shutdown; atomic_t usage_count;