aboutsummaryrefslogtreecommitdiffstats
path: root/sound
diff options
context:
space:
mode:
Diffstat (limited to 'sound')
-rw-r--r--sound/core/hwdep.c4
-rw-r--r--sound/core/oss/pcm_plugin.c20
-rw-r--r--sound/core/pcm_lib.c1
-rw-r--r--sound/core/pcm_native.c20
-rw-r--r--sound/core/rawmidi.c31
-rw-r--r--sound/firewire/amdtp-stream-trace.h3
-rw-r--r--sound/firewire/fireface/ff-protocol-latter.c12
-rw-r--r--sound/firewire/fireface/ff-stream.c10
-rw-r--r--sound/isa/es1688/es1688.c4
-rw-r--r--sound/isa/opti9xx/miro.c9
-rw-r--r--sound/isa/opti9xx/opti92x-ad1848.c9
-rw-r--r--sound/pci/hda/hda_intel.c12
-rw-r--r--sound/pci/hda/patch_hdmi.c6
-rw-r--r--sound/pci/hda/patch_realtek.c249
-rw-r--r--sound/pci/ice1712/ice1712.c3
-rw-r--r--sound/soc/codecs/hdac_hdmi.c6
-rw-r--r--sound/soc/codecs/max9867.c4
-rw-r--r--sound/soc/codecs/sgtl5000.c34
-rw-r--r--sound/soc/codecs/sgtl5000.h1
-rw-r--r--sound/soc/meson/axg-card.c4
-rw-r--r--sound/soc/sh/rcar/ssi.c11
-rw-r--r--sound/soc/sh/rcar/ssiu.c2
-rw-r--r--sound/soc/soc-topology.c115
-rw-r--r--sound/usb/card.c19
-rw-r--r--sound/usb/line6/podhd.c22
-rw-r--r--sound/usb/mixer.c8
-rw-r--r--sound/usb/mixer_maps.c24
-rw-r--r--sound/usb/quirks-table.h46
-rw-r--r--sound/usb/quirks.c11
-rw-r--r--sound/usb/usbaudio.h2
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;
generated by cgit 1.2.3-korg (git 2.39.0) at 2024-04-27 09:32:45 +0000