diff options
Diffstat (limited to 'sound')
30 files changed, 577 insertions, 125 deletions
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index b412d3b3d5ff..21edb8ac95eb 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -216,12 +216,12 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, if (info.index >= 32) return -EINVAL; /* check whether the dsp was already loaded */ - if (hw->dsp_loaded & (1 << info.index)) + if (hw->dsp_loaded & (1u << info.index)) return -EBUSY; err = hw->ops.dsp_load(hw, &info); if (err < 0) return err; - hw->dsp_loaded |= (1 << info.index); + hw->dsp_loaded |= (1u << info.index); return 0; } diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 50c35ecc8953..d1760f86773c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -211,21 +211,23 @@ static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug, if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_prev; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_first(plug); while (plugin && drv_frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -251,26 +253,28 @@ static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } } else diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 872a852de75c..d531e1bc2b81 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -433,6 +433,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, no_delta_check: if (runtime->status->hw_ptr == new_hw_ptr) { + runtime->hw_ptr_jiffies = curr_jiffies; update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return 0; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d5443eeb8b63..c936976e0e7b 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 @@ -2163,6 +2173,12 @@ 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; @@ -2191,7 +2207,7 @@ 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); @@ -2207,7 +2223,7 @@ 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/core/rawmidi.c b/sound/core/rawmidi.c index 20dd08e1f675..2a688b711a9a 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -120,6 +120,17 @@ static void snd_rawmidi_input_event_work(struct work_struct *work) runtime->event(runtime->substream); } +/* buffer refcount management: call with runtime->lock held */ +static inline void snd_rawmidi_buffer_ref(struct snd_rawmidi_runtime *runtime) +{ + runtime->buffer_ref++; +} + +static inline void snd_rawmidi_buffer_unref(struct snd_rawmidi_runtime *runtime) +{ + runtime->buffer_ref--; +} + static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; @@ -669,6 +680,11 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (!newbuf) return -ENOMEM; spin_lock_irq(&runtime->lock); + if (runtime->buffer_ref) { + spin_unlock_irq(&runtime->lock); + kvfree(newbuf); + return -EBUSY; + } oldbuf = runtime->buffer; runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; @@ -1019,8 +1035,10 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, long result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; unsigned long appl_ptr; + int err = 0; spin_lock_irqsave(&runtime->lock, flags); + snd_rawmidi_buffer_ref(runtime); while (count > 0 && runtime->avail) { count1 = runtime->buffer_size - runtime->appl_ptr; if (count1 > count) @@ -1039,16 +1057,19 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, if (userbuf) { spin_unlock_irqrestore(&runtime->lock, flags); if (copy_to_user(userbuf + result, - runtime->buffer + appl_ptr, count1)) { - return result > 0 ? result : -EFAULT; - } + runtime->buffer + appl_ptr, count1)) + err = -EFAULT; spin_lock_irqsave(&runtime->lock, flags); + if (err) + goto out; } result += count1; count -= count1; } + out: + snd_rawmidi_buffer_unref(runtime); spin_unlock_irqrestore(&runtime->lock, flags); - return result; + return result > 0 ? result : err; } long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream, @@ -1342,6 +1363,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, return -EAGAIN; } } + snd_rawmidi_buffer_ref(runtime); while (count > 0 && runtime->avail > 0) { count1 = runtime->buffer_size - runtime->appl_ptr; if (count1 > count) @@ -1373,6 +1395,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, } __end: count1 = runtime->avail < runtime->buffer_size; + snd_rawmidi_buffer_unref(runtime); spin_unlock_irqrestore(&runtime->lock, flags); if (count1) snd_rawmidi_output_trigger(substream, 1); diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 16c7f6605511..26e7cb555d3c 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -66,8 +66,7 @@ TRACE_EVENT(amdtp_packet, __entry->irq, __entry->index, __print_array(__get_dynamic_array(cip_header), - __get_dynamic_array_len(cip_header), - sizeof(u8))) + __get_dynamic_array_len(cip_header), 1)) ); #endif diff --git a/sound/firewire/fireface/ff-protocol-latter.c b/sound/firewire/fireface/ff-protocol-latter.c index 0e4c3a9ed5e4..76ae568489ef 100644 --- a/sound/firewire/fireface/ff-protocol-latter.c +++ b/sound/firewire/fireface/ff-protocol-latter.c @@ -107,18 +107,18 @@ static int latter_allocate_resources(struct snd_ff *ff, unsigned int rate) int err; // Set the number of data blocks transferred in a second. - if (rate % 32000 == 0) - code = 0x00; + if (rate % 48000 == 0) + code = 0x04; else if (rate % 44100 == 0) code = 0x02; - else if (rate % 48000 == 0) - code = 0x04; + else if (rate % 32000 == 0) + code = 0x00; else return -EINVAL; if (rate >= 64000 && rate < 128000) code |= 0x08; - else if (rate >= 128000 && rate < 192000) + else if (rate >= 128000) code |= 0x10; reg = cpu_to_le32(code); @@ -140,7 +140,7 @@ static int latter_allocate_resources(struct snd_ff *ff, unsigned int rate) if (curr_rate == rate) break; } - if (count == 10) + if (count > 10) return -ETIMEDOUT; for (i = 0; i < ARRAY_SIZE(amdtp_rate_table); ++i) { diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c index 63b79c4a5405..5452115c0ef9 100644 --- a/sound/firewire/fireface/ff-stream.c +++ b/sound/firewire/fireface/ff-stream.c @@ -184,7 +184,6 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) */ if (!amdtp_stream_running(&ff->rx_stream)) { int spd = fw_parent_device(ff->unit)->max_speed; - unsigned int ir_delay_cycle; err = ff->spec->protocol->begin_session(ff, rate); if (err < 0) @@ -200,14 +199,7 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) if (err < 0) goto error; - // The device postpones start of transmission mostly for several - // cycles after receiving packets firstly. - if (ff->spec->protocol == &snd_ff_protocol_ff800) - ir_delay_cycle = 800; // = 100 msec - else - ir_delay_cycle = 16; // = 2 msec - - err = amdtp_domain_start(&ff->domain, ir_delay_cycle); + err = amdtp_domain_start(&ff->domain, 0); if (err < 0) goto error; diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index ff3a05ad99c0..64610571a5e1 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -267,8 +267,10 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, return error; } error = snd_es1688_probe(card, dev); - if (error < 0) + if (error < 0) { + snd_card_free(card); return error; + } pnp_set_card_drvdata(pcard, card); snd_es968_pnp_is_probed = 1; return 0; diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index e764816a8f7a..b039429e6871 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -867,10 +867,13 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, spin_unlock_irqrestore(&chip->lock, flags); } +static inline void snd_miro_write_mask(struct snd_miro *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_miro_read(chip, reg); -#define snd_miro_write_mask(chip, reg, value, mask) \ - snd_miro_write(chip, reg, \ - (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) + snd_miro_write(chip, reg, (oldval & ~mask) | (value & mask)); +} /* * Proc Interface diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d06b29693c85..0e6d20e49158 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -317,10 +317,13 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, } -#define snd_opti9xx_write_mask(chip, reg, value, mask) \ - snd_opti9xx_write(chip, reg, \ - (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) +static inline void snd_opti9xx_write_mask(struct snd_opti9xx *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_opti9xx_read(chip, reg); + snd_opti9xx_write(chip, reg, (oldval & ~mask) | (value & mask)); +} static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 59b60b1f26f8..29da0b03b895 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2075,9 +2075,10 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * some HD-audio PCI entries are exposed without any codecs, and such devices * should be ignored from the beginning. */ -static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), - SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), +static const struct pci_device_id driver_blacklist[] = { + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */ {} }; @@ -2097,7 +2098,7 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + if (pci_match_id(driver_blacklist, pci)) { dev_info(&pci->dev, "Skipping the blacklisted device\n"); return -ENODEV; } @@ -2658,6 +2659,9 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab20), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + { PCI_DEVICE(0x1002, 0xab28), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8bc4d66ff986..0f3250417b95 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1934,8 +1934,10 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) /* Add sanity check to pass klockwork check. * This should never happen. */ - if (WARN_ON(spdif == NULL)) + if (WARN_ON(spdif == NULL)) { + mutex_unlock(&codec->spdif_mutex); return true; + } non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; @@ -2318,7 +2320,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + struct hdmi_eld *pin_eld = &per_pin->sink_eld; + pin_eld->eld_valid = false; hdmi_present_sense(per_pin, 0); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2fccf267b48..2c4575909441 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -86,6 +86,14 @@ struct alc_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; + unsigned int mute_led_coef_idx; + unsigned int mute_led_coefbit_mask; + unsigned int mute_led_coefbit_on; + unsigned int mute_led_coefbit_off; + unsigned int mic_led_coef_idx; + unsigned int mic_led_coefbit_mask; + unsigned int mic_led_coefbit_on; + unsigned int mic_led_coefbit_off; hda_nid_t headset_mic_pin; hda_nid_t headphone_mic_pin; @@ -376,6 +384,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: + case 0x10ec0287: case 0x10ec0288: case 0x10ec0285: case 0x10ec0298: @@ -2449,6 +2458,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), @@ -2464,6 +2474,9 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), @@ -4182,6 +4195,111 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } +/* update mute-LED according to the speaker mute state via COEF bit */ +static void alc_fixup_mute_led_coefbit_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct alc_spec *spec = codec->spec; + + if (spec->mute_led_polarity) + enabled = !enabled; + + /* temporarily power up/down for setting COEF bit */ + enabled ? alc_update_coef_idx(codec, spec->mute_led_coef_idx, + spec->mute_led_coefbit_mask, spec->mute_led_coefbit_off) : + alc_update_coef_idx(codec, spec->mute_led_coef_idx, + spec->mute_led_coefbit_mask, spec->mute_led_coefbit_on); +} + +static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef_idx = 0x0b; + spec->mute_led_coefbit_mask = 1<<3; + spec->mute_led_coefbit_on = 1<<3; + spec->mute_led_coefbit_off = 0; + spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook; + spec->gen.vmaster_mute_enum = 1; + } +} + +static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef_idx = 0x34; + spec->mute_led_coefbit_mask = 1<<5; + spec->mute_led_coefbit_on = 0; + spec->mute_led_coefbit_off = 1<<5; + spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook; + spec->gen.vmaster_mute_enum = 1; + } +} + +/* turn on/off mic-mute LED per capture hook by coef bit */ +static void alc_hp_cap_micmute_update(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec->gen.micmute_led.led_value) + alc_update_coef_idx(codec, spec->mic_led_coef_idx, + spec->mic_led_coefbit_mask, spec->mic_led_coefbit_on); + else + alc_update_coef_idx(codec, spec->mic_led_coef_idx, + spec->mic_led_coefbit_mask, spec->mic_led_coefbit_off); +} + +static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mic_led_coef_idx = 0x19; + spec->mic_led_coefbit_mask = 1<<13; + spec->mic_led_coefbit_on = 1<<13; + spec->mic_led_coefbit_off = 0; + snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + } +} + +static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mic_led_coef_idx = 0x35; + spec->mic_led_coefbit_mask = 3<<2; + spec->mic_led_coefbit_on = 2<<2; + spec->mic_led_coefbit_off = 1<<2; + snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + } +} + +static void alc285_fixup_hp_mute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc285_fixup_hp_mute_led_coefbit(codec, fix, action); + alc285_fixup_hp_coef_micmute_led(codec, fix, action); +} + +static void alc236_fixup_hp_mute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc236_fixup_hp_mute_led_coefbit(codec, fix, action); + alc236_fixup_hp_coef_micmute_led(codec, fix, action); +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -5367,18 +5485,9 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, { 0x19, 0x21a11010 }, /* dock mic */ { } }; - /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise - * the speaker output becomes too low by some reason on Thinkpads with - * ALC298 codec - */ - static const hda_nid_t preferred_pairs[] = { - 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, - 0 - }; struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.preferred_dacs = preferred_pairs; spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; snd_hda_apply_pincfgs(codec, pincfgs); } else if (action == HDA_FIXUP_ACT_INIT) { @@ -5391,6 +5500,23 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, } } +static void alc_fixup_tpt470_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise + * the speaker output becomes too low by some reason on Thinkpads with + * ALC298 codec + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.preferred_dacs = preferred_pairs; +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5743,6 +5869,15 @@ static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, } } +static void alc225_fixup_s3_pop_noise(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + codec->power_save_node = 1; +} + /* Forcibly assign NID 0x03 to HP/LO while NID 0x02 to SPK for EQ */ static void alc274_fixup_bind_dacs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5847,6 +5982,7 @@ enum { ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST, ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, @@ -5932,9 +6068,11 @@ enum { ALC233_FIXUP_ACER_HEADSET_MIC, ALC294_FIXUP_LENOVO_MIC_LOCATION, ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, + ALC225_FIXUP_S3_POP_NOISE, ALC700_FIXUP_INTEL_REFERENCE, ALC274_FIXUP_DELL_BIND_DACS, ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + ALC298_FIXUP_TPT470_DOCK_FIX, ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, @@ -5967,7 +6105,12 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, + ALC285_FIXUP_HP_MUTE_LED, + ALC236_FIXUP_HP_MUTE_LED, + ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, + ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6165,6 +6308,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT }, + [ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_LENOVO_DOCK, + }, [ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, @@ -6817,6 +6966,12 @@ static const struct hda_fixup alc269_fixups[] = { { } }, .chained = true, + .chain_id = ALC225_FIXUP_S3_POP_NOISE + }, + [ALC225_FIXUP_S3_POP_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc225_fixup_s3_pop_noise, + .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, [ALC700_FIXUP_INTEL_REFERENCE] = { @@ -6849,12 +7004,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC274_FIXUP_DELL_BIND_DACS }, - [ALC298_FIXUP_TPT470_DOCK] = { + [ALC298_FIXUP_TPT470_DOCK_FIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_tpt470_dock, .chained = true, .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE }, + [ALC298_FIXUP_TPT470_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt470_dacs, + .chained = true, + .chain_id = ALC298_FIXUP_TPT470_DOCK_FIX + }, [ALC255_FIXUP_DUMMY_LINEOUT_VERB] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7089,10 +7250,45 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_COEF_1B] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set bit 10 to correct noisy output after reboot from + * Windows 10 (due to pop noise reduction?) + */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x1b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b }, + { } + }, + }, [ALC285_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_gpio_led, }, + [ALC285_FIXUP_HP_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_mute_led, + }, + [ALC236_FIXUP_HP_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc236_fixup_hp_mute_led, + }, + [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc5 }, + { } + }, + }, + [ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7238,6 +7434,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -7258,8 +7456,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -7285,6 +7485,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), 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), @@ -7295,12 +7497,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), 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, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST), SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), @@ -7439,6 +7642,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HEADSET_MODE, .name = "headset-mode"}, {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, + {.id = ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST, .name = "lenovo-dock-limit-boost"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, @@ -7450,6 +7654,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, + {.id = ALC298_FIXUP_TPT470_DOCK_FIX, .name = "tpt470-dock-fix"}, {.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, @@ -7867,6 +8072,18 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x17, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60120}, + {0x17, 0x90170110}, + {0x21, 0x04211030}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x04211020}), @@ -7907,6 +8124,12 @@ 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}), {} }; @@ -8076,6 +8299,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0245: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0289: spec->codec_variant = ALC269_TYPE_ALC215; spec->shutup = alc225_shutup; @@ -8083,8 +8307,6 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: - codec->power_save_node = 1; - /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; @@ -9356,6 +9578,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269), HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269), HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0287, "ALC287", patch_alc269), HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269), HDA_CODEC_ENTRY(0x10ec0289, "ALC289", patch_alc269), HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 884d0cdec08c..73e1e5400506 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2332,7 +2332,8 @@ static int snd_ice1712_chip_init(struct snd_ice1712 *ice) pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]); pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]); pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]); - if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24) { + if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24 && + ice->eeprom.subvendor != ICE1712_SUBDEVICE_STAUDIO_ADCIII) { ice->gpio.write_mask = ice->eeprom.gpiomask; ice->gpio.direction = ice->eeprom.gpiodir; snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index e6558475e006..f0f689ddbefe 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { - struct hdac_hdmi_pcm *pcm = NULL; + struct hdac_hdmi_pcm *pcm; list_for_each_entry(pcm, &hdmi->pcm_list, head) { if (pcm->cvt == cvt) - break; + return pcm; } - return pcm; + return NULL; } static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 8600c5439e1e..2e4aa23b5a60 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -46,13 +46,13 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, static const struct snd_kcontrol_new max9867_snd_controls[] = { SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + MAX9867_RIGHTVOL, 0, 40, 1, max9867_master_tlv), SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + MAX9867_RIGHTMICGAIN, 5, 3, 0, max9867_micboost_tlv), SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, max9867_dac_tlv), diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d5130193b4a2..e8a8bf7b4ffe 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "Error %d initializing CHIP_CLK_CTRL\n", ret); + /* Mute everything to avoid pop from the following power-up */ + ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_CHIP_ANA_CTRL_DEFAULT); + if (ret) { + dev_err(&client->dev, + "Error %d muting outputs via CHIP_ANA_CTRL\n", ret); + goto disable_clk; + } + + /* + * If VAG is powered-on (e.g. from previous boot), it would be disabled + * by the write to ANA_POWER in later steps of the probe code. This + * may create a loud pop even with all outputs muted. The proper way + * to circumvent this is disabling the bit first and waiting the proper + * cool-down time. + */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value); + if (ret) { + dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret); + goto disable_clk; + } + if (value & SGTL5000_VAG_POWERUP) { + ret = regmap_update_bits(sgtl5000->regmap, + SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, + 0); + if (ret) { + dev_err(&client->dev, "Error %d disabling VAG\n", ret); + goto disable_clk; + } + + msleep(SGTL5000_VAG_POWERDOWN_DELAY); + } + /* Follow section 2.2.1.1 of AN3663 */ ana_pwr = SGTL5000_ANA_POWER_DEFAULT; if (sgtl5000->num_supplies <= VDDD) { diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index a4bf4bca95bf..56ec5863f250 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -233,6 +233,7 @@ /* * SGTL5000_CHIP_ANA_CTRL */ +#define SGTL5000_CHIP_ANA_CTRL_DEFAULT 0x0133 #define SGTL5000_LINE_OUT_MUTE 0x0100 #define SGTL5000_HP_SEL_MASK 0x0040 #define SGTL5000_HP_SEL_SHIFT 6 diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 2b04ac3d8fd3..1f698adde506 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -586,10 +586,8 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) ret = axg_card_parse_tdm(card, np, index); - else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) { + else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) dai_link->params = &codec_params; - dai_link->no_pcm = 0; /* link is not a DPCM BE */ - } return ret; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fc5d089868df..4a7d3413917f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -594,10 +594,16 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod, * Capture: It might not receave data. Do nothing */ if (rsnd_io_is_play(io)) { - rsnd_mod_write(mod, SSICR, cr | EN); + rsnd_mod_write(mod, SSICR, cr | ssi->cr_en); rsnd_ssi_status_check(mod, DIRQ); } + /* In multi-SSI mode, stop is performed by setting ssi0129 in + * SSI_CONTROL to 0 (in rsnd_ssio_stop_gen2). Do nothing here. + */ + if (rsnd_ssi_multi_slaves_runtime(io)) + return 0; + /* * disable SSI, * and, wait idle state @@ -737,6 +743,9 @@ static void rsnd_ssi_parent_attach(struct rsnd_mod *mod, if (!rsnd_rdai_is_clk_master(rdai)) return; + if (rsnd_ssi_is_multi_slave(mod, io)) + return; + switch (rsnd_mod_id(mod)) { case 1: case 2: diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index f35d88211887..9c7c3e7539c9 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -221,7 +221,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, i; for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) { - shift = (i * 4) + 16; + shift = (i * 4) + 20; val = (val & ~(0xF << shift)) | rsnd_mod_id(pos) << shift; } diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index a152409e8746..009d65a6fb43 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -894,7 +894,13 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + err = soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + kfree(sm); + continue; + } /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, @@ -1118,6 +1124,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_ctl_hdr *control_hdr; + int ret; int i; if (tplg->pass != SOC_TPLG_PASS_MIXER) { @@ -1146,25 +1153,30 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: case SND_SOC_TPLG_DAPM_CTL_PIN: - soc_tplg_dmixer_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_dmixer_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_CTL_ENUM_VALUE: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - soc_tplg_denum_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_denum_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_BYTES: - soc_tplg_dbytes_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_dbytes_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; default: soc_bind_err(tplg, control_hdr, i); return -EINVAL; } + if (ret < 0) { + dev_err(tplg->dev, "ASoC: invalid control\n"); + return ret; + } + } return 0; @@ -1272,7 +1284,9 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, routes[i]->dobj.index = tplg->index; list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list); - soc_tplg_add_route(tplg, routes[i]); + ret = soc_tplg_add_route(tplg, routes[i]); + if (ret < 0) + break; /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, routes[i], 1); @@ -1355,7 +1369,13 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + kfree(sm); + continue; + } /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], @@ -1766,10 +1786,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1800,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1837,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1869,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1860,6 +1890,14 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, return ret; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1954,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2088,7 +2135,9 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, _pcm = pcm; } else { abi_match = false; - pcm_new_ver(tplg, pcm, &_pcm); + ret = pcm_new_ver(tplg, pcm, &_pcm); + if (ret < 0) + return ret; } /* create the FE DAIs and DAI links */ @@ -2436,13 +2485,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2507,15 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + return ret; } /* load physical DAI elements */ @@ -2466,7 +2524,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, { struct snd_soc_tplg_dai *dai; int count; - int i; + int i, ret; count = le32_to_cpu(hdr->count); @@ -2481,7 +2539,12 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, return -EINVAL; } - soc_tplg_dai_config(tplg, dai); + ret = soc_tplg_dai_config(tplg, dai); + if (ret < 0) { + dev_err(tplg->dev, "ASoC: failed to configure DAI\n"); + return ret; + } + tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size)); } @@ -2589,7 +2652,7 @@ static int soc_valid_header(struct soc_tplg *tplg, } /* big endian firmware objects not supported atm */ - if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) { + if (le32_to_cpu(hdr->magic) == SOC_TPLG_MAGIC_BIG_ENDIAN) { dev_err(tplg->dev, "ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n", tplg->pass, hdr->magic, diff --git a/sound/usb/card.c b/sound/usb/card.c index 827fb0bc8b56..8f559b505bb7 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -813,9 +813,6 @@ 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); @@ -828,6 +825,11 @@ 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; } @@ -841,10 +843,10 @@ 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); @@ -866,9 +868,12 @@ 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/line6/podhd.c b/sound/usb/line6/podhd.c index d37db32ecd3b..e39dc85c355a 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -21,8 +21,7 @@ enum { LINE6_PODHD300, LINE6_PODHD400, - LINE6_PODHD500_0, - LINE6_PODHD500_1, + LINE6_PODHD500, LINE6_PODX3, LINE6_PODX3LIVE, LINE6_PODHD500X, @@ -318,8 +317,7 @@ static const struct usb_device_id podhd_id_table[] = { /* TODO: no need to alloc data interfaces when only audio is used */ { LINE6_DEVICE(0x5057), .driver_info = LINE6_PODHD300 }, { LINE6_DEVICE(0x5058), .driver_info = LINE6_PODHD400 }, - { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500_0 }, - { LINE6_IF_NUM(0x414D, 1), .driver_info = LINE6_PODHD500_1 }, + { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500 }, { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, @@ -352,23 +350,13 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x82, .ep_audio_w = 0x01, }, - [LINE6_PODHD500_0] = { + [LINE6_PODHD500] = { .id = "PODHD500", .name = "POD HD500", - .capabilities = LINE6_CAP_PCM + .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL | LINE6_CAP_HWMON, .altsetting = 1, - .ep_ctrl_r = 0x81, - .ep_ctrl_w = 0x01, - .ep_audio_r = 0x86, - .ep_audio_w = 0x02, - }, - [LINE6_PODHD500_1] = { - .id = "PODHD500", - .name = "POD HD500", - .capabilities = LINE6_CAP_PCM - | LINE6_CAP_HWMON, - .altsetting = 0, + .ctrl_if = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7a2961ad60de..68fefe55e5c0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1171,6 +1171,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; + case USB_ID(0x0495, 0x3042): /* ESS Technology Asus USB DAC */ + if ((strstr(kctl->id.name, "Playback Volume") != NULL) || + strstr(kctl->id.name, "Capture Volume") != NULL) { + cval->min >>= 8; + cval->max = 0; + cval->res = 1; + } + break; } } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0260c750e156..9af7aa93f6fa 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -397,6 +397,21 @@ static const struct usbmix_connector_map trx40_mobo_connector_map[] = { {} }; +/* Rear panel + front mic on Gigabyte TRX40 Aorus Master with ALC1220-VB */ +static const struct usbmix_name_map aorus_master_alc1220vb_map[] = { + { 17, NULL }, /* OT, IEC958?, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Line Out" }, /* OT */ + { 22, "Line Out Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + {} +}; + /* * Control map entries */ @@ -526,6 +541,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1b1c, 0x0a42), .map = corsair_virtuoso_map, }, + { /* Gigabyte TRX40 Aorus Master (rear panel + front mic) */ + .id = USB_ID(0x0414, 0xa001), + .map = aorus_master_alc1220vb_map, + }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), .map = trx40_mobo_map, @@ -549,6 +568,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Asrock TRX40 Creator */ + .id = USB_ID(0x26ce, 0x0a01), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8c2f5c23e1b4..042a5e8eb79d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -25,6 +25,26 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* HP Thunderbolt Dock Audio Headset */ +{ + USB_DEVICE(0x03f0, 0x0269), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "HP", + .product_name = "Thunderbolt Dock Audio Headset", + .profile_name = "HP-Thunderbolt-Dock-Audio-Headset", + .ifnum = QUIRK_NO_INTERFACE + } +}, +/* HP Thunderbolt Dock Audio Module */ +{ + USB_DEVICE(0x03f0, 0x0567), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "HP", + .product_name = "Thunderbolt Dock Audio Module", + .profile_name = "HP-Thunderbolt-Dock-Audio-Module", + .ifnum = QUIRK_NO_INTERFACE + } +}, /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), @@ -3647,6 +3667,32 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP +/* Two entries for Gigabyte TRX40 Aorus Master: + * TRX40 Aorus Master has two USB-audio devices, one for the front headphone + * with ESS SABRE9218 DAC chip, while another for the rest I/O (the rear + * panel and the front mic) with Realtek ALC1220-VB. + * Here we provide two distinct names for making UCM profiles easier. + */ +{ + USB_DEVICE(0x0414, 0xa000), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Gigabyte", + .product_name = "Aorus Master Front Headphone", + .profile_name = "Gigabyte-Aorus-Master-Front-Headphone", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ + USB_DEVICE(0x0414, 0xa001), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Gigabyte", + .product_name = "Aorus Master Main Audio", + .profile_name = "Gigabyte-Aorus-Master-Main-Audio", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7f558f4b4520..732580bdc6a4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1592,13 +1592,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); - /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, - * otherwise requests like get/set frequency return as failed despite - * actually succeeding. + /* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny + * delay here, otherwise requests like get/set frequency return as + * failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || chip->usb_id == USB_ID(0x046d, 0x0a46) || - chip->usb_id == USB_ID(0x0b0e, 0x0349)) && + chip->usb_id == USB_ID(0x0b0e, 0x0349) || + chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) usleep_range(1000, 2000); } @@ -1643,7 +1644,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ - case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ + case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 6fe3ab582ec6..a42d021624dc 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -26,7 +26,7 @@ struct snd_usb_audio { struct usb_interface *pm_intf; u32 usb_id; struct mutex mutex; - unsigned int autosuspended:1; + unsigned int system_suspend; atomic_t active; atomic_t shutdown; atomic_t usage_count; |