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-rw-r--r--include/sound/compress_driver.h10
-rw-r--r--sound/core/compress_offload.c4
-rw-r--r--sound/drivers/opl3/opl3_synth.c2
-rw-r--r--sound/pci/hda/hda_auto_parser.c6
-rw-r--r--sound/pci/hda/patch_hdmi.c41
-rw-r--r--sound/pci/hda/patch_realtek.c38
-rw-r--r--sound/soc/amd/renoir/Makefile7
-rw-r--r--sound/soc/codecs/rt5682.c9
-rw-r--r--sound/soc/fsl/fsl_mqs.c23
-rw-r--r--sound/usb/card.h6
-rw-r--r--sound/usb/endpoint.c18
-rw-r--r--sound/usb/pcm.c1
-rw-r--r--sound/usb/quirks-table.h52
13 files changed, 174 insertions, 43 deletions
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index 6ce8effa0b12..70cbc5095e72 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -66,6 +66,7 @@ struct snd_compr_runtime {
* @direction: stream direction, playback/recording
* @metadata_set: metadata set flag, true when set
* @next_track: has userspace signal next track transition, true when set
+ * @partial_drain: undergoing partial_drain for stream, true when set
* @private_data: pointer to DSP private data
* @dma_buffer: allocated buffer if any
*/
@@ -78,6 +79,7 @@ struct snd_compr_stream {
enum snd_compr_direction direction;
bool metadata_set;
bool next_track;
+ bool partial_drain;
void *private_data;
struct snd_dma_buffer dma_buffer;
};
@@ -182,7 +184,13 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream)
if (snd_BUG_ON(!stream))
return;
- stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+ /* for partial_drain case we are back to running state on success */
+ if (stream->partial_drain) {
+ stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
+ stream->partial_drain = false; /* clear this flag as well */
+ } else {
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+ }
wake_up(&stream->runtime->sleep);
}
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 509290f2efa8..0e53f6f31916 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -764,6 +764,9 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
if (!retval) {
+ /* clear flags and stop any drain wait */
+ stream->partial_drain = false;
+ stream->metadata_set = false;
snd_compr_drain_notify(stream);
stream->runtime->total_bytes_available = 0;
stream->runtime->total_bytes_transferred = 0;
@@ -921,6 +924,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream)
if (stream->next_track == false)
return -EPERM;
+ stream->partial_drain = true;
retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN);
if (retval) {
pr_debug("Partial drain returned failure\n");
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index e69a4ef0d6bd..08c10ac9d6c8 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -91,6 +91,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
{
struct snd_dm_fm_info info;
+ memset(&info, 0, sizeof(info));
+
info.fm_mode = opl3->fm_mode;
info.rhythm = opl3->rhythm;
if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info)))
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 2c6d2becfe1a..824f4ac1a8ce 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -72,6 +72,12 @@ static int compare_input_type(const void *ap, const void *bp)
if (a->type != b->type)
return (int)(a->type - b->type);
+ /* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */
+ if (a->is_headset_mic && b->is_headphone_mic)
+ return -1; /* don't swap */
+ else if (a->is_headphone_mic && b->is_headset_mic)
+ return 1; /* swap */
+
/* In case one has boost and the other one has not,
pick the one with boost first. */
return (int)(b->has_boost_on_pin - a->has_boost_on_pin);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index e2b21ef5d7d1..41eaa89660c3 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -259,7 +259,7 @@ static int hinfo_to_pcm_index(struct hda_codec *codec,
if (get_pcm_rec(spec, pcm_idx)->stream == hinfo)
return pcm_idx;
- codec_warn(codec, "HDMI: hinfo %p not registered\n", hinfo);
+ codec_warn(codec, "HDMI: hinfo %p not tied to a PCM\n", hinfo);
return -EINVAL;
}
@@ -277,7 +277,8 @@ static int hinfo_to_pin_index(struct hda_codec *codec,
return pin_idx;
}
- codec_dbg(codec, "HDMI: hinfo %p not registered\n", hinfo);
+ codec_dbg(codec, "HDMI: hinfo %p (pcm %d) not registered\n", hinfo,
+ hinfo_to_pcm_index(codec, hinfo));
return -EINVAL;
}
@@ -1804,33 +1805,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
static int hdmi_parse_codec(struct hda_codec *codec)
{
- hda_nid_t nid;
+ hda_nid_t start_nid;
+ unsigned int caps;
int i, nodes;
- nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid);
- if (!nid || nodes < 0) {
+ nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid);
+ if (!start_nid || nodes < 0) {
codec_warn(codec, "HDMI: failed to get afg sub nodes\n");
return -EINVAL;
}
- for (i = 0; i < nodes; i++, nid++) {
- unsigned int caps;
- unsigned int type;
+ /*
+ * hdmi_add_pin() assumes total amount of converters to
+ * be known, so first discover all converters
+ */
+ for (i = 0; i < nodes; i++) {
+ hda_nid_t nid = start_nid + i;
caps = get_wcaps(codec, nid);
- type = get_wcaps_type(caps);
if (!(caps & AC_WCAP_DIGITAL))
continue;
- switch (type) {
- case AC_WID_AUD_OUT:
+ if (get_wcaps_type(caps) == AC_WID_AUD_OUT)
hdmi_add_cvt(codec, nid);
- break;
- case AC_WID_PIN:
+ }
+
+ /* discover audio pins */
+ for (i = 0; i < nodes; i++) {
+ hda_nid_t nid = start_nid + i;
+
+ caps = get_wcaps(codec, nid);
+
+ if (!(caps & AC_WCAP_DIGITAL))
+ continue;
+
+ if (get_wcaps_type(caps) == AC_WID_PIN)
hdmi_add_pin(codec, nid);
- break;
- }
}
return 0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 737ef82a75fd..194ffa8c66ce 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6149,6 +6149,9 @@ enum {
ALC236_FIXUP_HP_MUTE_LED,
ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+ ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS,
+ ALC269VC_FIXUP_ACER_HEADSET_MIC,
+ ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -7327,6 +7330,35 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE
},
+ [ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x14, 0x90100120 }, /* use as internal speaker */
+ { 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */
+ { 0x1a, 0x01011020 }, /* use as line out */
+ { },
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC269VC_FIXUP_ACER_HEADSET_MIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x02a11030 }, /* use as headset mic */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x01a11130 }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7342,10 +7374,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK),
+ SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS),
+ SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
@@ -7571,8 +7606,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
- SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
- SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+ SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
diff --git a/sound/soc/amd/renoir/Makefile b/sound/soc/amd/renoir/Makefile
index e4371932a55a..4a82690aec16 100644
--- a/sound/soc/amd/renoir/Makefile
+++ b/sound/soc/amd/renoir/Makefile
@@ -2,6 +2,7 @@
# Renoir platform Support
snd-rn-pci-acp3x-objs := rn-pci-acp3x.o
snd-acp3x-pdm-dma-objs := acp3x-pdm-dma.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-rn-pci-acp3x.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-acp3x-pdm-dma.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH) += acp3x-rn.o
+snd-acp3x-rn-objs := acp3x-rn.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-rn-pci-acp3x.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-acp3x-pdm-dma.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH) += snd-acp3x-rn.o
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 3e9d2c6c51f9..7d6670abdb08 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -932,7 +932,9 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
RT5682_PWR_CBJ, RT5682_PWR_CBJ);
-
+ snd_soc_component_update_bits(component,
+ RT5682_HP_CHARGE_PUMP_1,
+ RT5682_OSW_L_MASK | RT5682_OSW_R_MASK, 0);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH);
@@ -956,6 +958,11 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
rt5682->jack_type = SND_JACK_HEADPHONE;
break;
}
+
+ snd_soc_component_update_bits(component,
+ RT5682_HP_CHARGE_PUMP_1,
+ RT5682_OSW_L_MASK | RT5682_OSW_R_MASK,
+ RT5682_OSW_L_EN | RT5682_OSW_R_EN);
} else {
rt5682_enable_push_button_irq(component, false);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c
index 0c813a45bba7..69aeb0e71844 100644
--- a/sound/soc/fsl/fsl_mqs.c
+++ b/sound/soc/fsl/fsl_mqs.c
@@ -265,12 +265,20 @@ static int fsl_mqs_remove(struct platform_device *pdev)
static int fsl_mqs_runtime_resume(struct device *dev)
{
struct fsl_mqs *mqs_priv = dev_get_drvdata(dev);
+ int ret;
- if (mqs_priv->ipg)
- clk_prepare_enable(mqs_priv->ipg);
+ ret = clk_prepare_enable(mqs_priv->ipg);
+ if (ret) {
+ dev_err(dev, "failed to enable ipg clock\n");
+ return ret;
+ }
- if (mqs_priv->mclk)
- clk_prepare_enable(mqs_priv->mclk);
+ ret = clk_prepare_enable(mqs_priv->mclk);
+ if (ret) {
+ dev_err(dev, "failed to enable mclk clock\n");
+ clk_disable_unprepare(mqs_priv->ipg);
+ return ret;
+ }
if (mqs_priv->use_gpr)
regmap_write(mqs_priv->regmap, IOMUXC_GPR2,
@@ -292,11 +300,8 @@ static int fsl_mqs_runtime_suspend(struct device *dev)
regmap_read(mqs_priv->regmap, REG_MQS_CTRL,
&mqs_priv->reg_mqs_ctrl);
- if (mqs_priv->mclk)
- clk_disable_unprepare(mqs_priv->mclk);
-
- if (mqs_priv->ipg)
- clk_disable_unprepare(mqs_priv->ipg);
+ clk_disable_unprepare(mqs_priv->mclk);
+ clk_disable_unprepare(mqs_priv->ipg);
return 0;
}
diff --git a/sound/usb/card.h b/sound/usb/card.h
index d6219fba9699..de43267b9c8a 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -84,10 +84,10 @@ struct snd_usb_endpoint {
dma_addr_t sync_dma; /* DMA address of syncbuf */
unsigned int pipe; /* the data i/o pipe */
- unsigned int framesize[2]; /* small/large frame sizes in samples */
- unsigned int sample_rem; /* remainder from division fs/fps */
+ unsigned int packsize[2]; /* small/large packet sizes in samples */
+ unsigned int sample_rem; /* remainder from division fs/pps */
unsigned int sample_accum; /* sample accumulator */
- unsigned int fps; /* frames per second */
+ unsigned int pps; /* packets per second */
unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */
unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */
int freqshift; /* how much to shift the feedback value to get Q16.16 */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 9bea7d3f99f8..88760268fb55 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -159,11 +159,11 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
return ep->maxframesize;
ep->sample_accum += ep->sample_rem;
- if (ep->sample_accum >= ep->fps) {
- ep->sample_accum -= ep->fps;
- ret = ep->framesize[1];
+ if (ep->sample_accum >= ep->pps) {
+ ep->sample_accum -= ep->pps;
+ ret = ep->packsize[1];
} else {
- ret = ep->framesize[0];
+ ret = ep->packsize[0];
}
return ret;
@@ -1088,15 +1088,15 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) {
ep->freqn = get_usb_full_speed_rate(rate);
- ep->fps = 1000;
+ ep->pps = 1000 >> ep->datainterval;
} else {
ep->freqn = get_usb_high_speed_rate(rate);
- ep->fps = 8000;
+ ep->pps = 8000 >> ep->datainterval;
}
- ep->sample_rem = rate % ep->fps;
- ep->framesize[0] = rate / ep->fps;
- ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps;
+ ep->sample_rem = rate % ep->pps;
+ ep->packsize[0] = rate / ep->pps;
+ ep->packsize[1] = (rate + (ep->pps - 1)) / ep->pps;
/* calculate the frequency in 16.16 format */
ep->freqm = ep->freqn;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a777d36c4f5a..40b7cd13fed9 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -368,6 +368,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
goto add_sync_ep_from_ifnum;
case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+ case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
ep = 0x81;
ifnum = 2;
goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 4ec491011b19..9092cc0aa807 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3633,4 +3633,56 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
}
},
+/*
+ * MacroSilicon MS2109 based HDMI capture cards
+ *
+ * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch.
+ * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if
+ * they pretend to be 96kHz mono as a workaround for stereo being broken
+ * by that...
+ *
+ * They also have swapped L-R channels, but that's for userspace to deal
+ * with.
+ */
+{
+ USB_DEVICE(0x534d, 0x2109),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "MacroSilicon",
+ .product_name = "MS2109",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = &(const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_ALIGN_TRANSFER,
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_MIXER,
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels = 2,
+ .iface = 3,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC |
+ USB_ENDPOINT_SYNC_ASYNC,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+
#undef USB_DEVICE_VENDOR_SPEC
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-22 15:19:30 +0000