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-rw-r--r--MAINTAINERS1
-rw-r--r--include/sound/pcm_params.h4
-rw-r--r--include/sound/soc.h2
-rw-r--r--sound/core/pcm_native.c14
-rw-r--r--sound/pci/hda/hda_intel.c4
-rw-r--r--sound/pci/hda/patch_realtek.c95
-rw-r--r--sound/soc/codecs/hdac_hdmi.c11
-rw-r--r--sound/soc/codecs/pcm186x.h2
-rw-r--r--sound/soc/codecs/pcm3060.c12
-rw-r--r--sound/soc/codecs/wm_adsp.c37
-rw-r--r--sound/soc/intel/Kconfig26
-rw-r--r--sound/soc/intel/boards/Kconfig24
-rw-r--r--sound/soc/intel/boards/cht_bsw_max98090_ti.c32
-rw-r--r--sound/soc/intel/skylake/skl.c32
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c67
-rw-r--r--sound/soc/omap/omap-dmic.c9
-rw-r--r--sound/soc/omap/omap-mcbsp.c6
-rw-r--r--sound/soc/omap/omap-mcpdm.c43
-rw-r--r--sound/soc/qcom/common.c9
-rw-r--r--sound/soc/qcom/qdsp6/q6afe-dai.c208
-rw-r--r--sound/soc/qcom/qdsp6/q6afe.c16
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c33
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c19
-rw-r--r--sound/soc/rockchip/rockchip_pcm.c1
-rw-r--r--sound/soc/sh/rcar/ssi.c2
-rw-r--r--sound/soc/soc-acpi.c10
-rw-r--r--sound/soc/soc-core.c1
-rw-r--r--sound/soc/stm/stm32_sai_sub.c2
-rw-r--r--sound/soc/sunxi/Kconfig2
-rw-r--r--sound/soc/sunxi/sun8i-codec.c12
-rw-r--r--sound/usb/card.c5
-rw-r--r--sound/usb/quirks-table.h10
-rw-r--r--sound/usb/quirks.c1
33 files changed, 485 insertions, 267 deletions
diff --git a/MAINTAINERS b/MAINTAINERS
index b755a89fa325..dfc90628092e 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -13931,6 +13931,7 @@ S: Supported
F: Documentation/devicetree/bindings/sound/
F: Documentation/sound/soc/
F: sound/soc/
+F: include/dt-bindings/sound/
F: include/sound/soc*
SOUNDWIRE SUBSYSTEM
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index 2dd37cada7c0..888a833d3b00 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -254,11 +254,13 @@ static inline int snd_interval_empty(const struct snd_interval *i)
static inline int snd_interval_single(const struct snd_interval *i)
{
return (i->min == i->max ||
- (i->min + 1 == i->max && i->openmax));
+ (i->min + 1 == i->max && (i->openmin || i->openmax)));
}
static inline int snd_interval_value(const struct snd_interval *i)
{
+ if (i->openmin && !i->openmax)
+ return i->max;
return i->min;
}
diff --git a/include/sound/soc.h b/include/sound/soc.h
index f1dab1f4b194..70c10a8f3e90 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1192,7 +1192,7 @@ struct snd_soc_pcm_runtime {
((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
(i)++)
#define for_each_rtd_codec_dai_rollback(rtd, i, dai) \
- for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);)
+ for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);)
/* mixer control */
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 66c90f486af9..818dff1de545 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -36,6 +36,7 @@
#include <sound/timer.h>
#include <sound/minors.h>
#include <linux/uio.h>
+#include <linux/delay.h>
#include "pcm_local.h"
@@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem);
* and this may lead to a deadlock when the code path takes read sem
* twice (e.g. one in snd_pcm_action_nonatomic() and another in
* snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to
- * spin until it gets the lock.
+ * sleep until all the readers are completed without blocking by writer.
*/
-static inline void down_write_nonblock(struct rw_semaphore *lock)
+static inline void down_write_nonfifo(struct rw_semaphore *lock)
{
while (!down_write_trylock(lock))
- cond_resched();
+ msleep(1);
}
#define PCM_LOCK_DEFAULT 0
@@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
res = -ENOMEM;
goto _nolock;
}
- down_write_nonblock(&snd_pcm_link_rwsem);
+ down_write_nonfifo(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
substream->runtime->status->state != substream1->runtime->status->state ||
@@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
struct snd_pcm_substream *s;
int res = 0;
- down_write_nonblock(&snd_pcm_link_rwsem);
+ down_write_nonfifo(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (!snd_pcm_stream_linked(substream)) {
res = -EALREADY;
@@ -2369,7 +2370,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
static void pcm_release_private(struct snd_pcm_substream *substream)
{
- snd_pcm_unlink(substream);
+ if (snd_pcm_stream_linked(substream))
+ snd_pcm_unlink(substream);
}
void snd_pcm_release_substream(struct snd_pcm_substream *substream)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0bbdf1a01e76..76f03abd15ab 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2498,6 +2498,10 @@ static const struct pci_device_id azx_ids[] = {
/* AMD Hudson */
{ PCI_DEVICE(0x1022, 0x780d),
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+ /* AMD Stoney */
+ { PCI_DEVICE(0x1022, 0x157a),
+ .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
+ AZX_DCAPS_PM_RUNTIME },
/* AMD Raven */
{ PCI_DEVICE(0x1022, 0x15e3),
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index db3cbdd6151f..8933441c2515 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -388,6 +388,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0285:
case 0x10ec0298:
case 0x10ec0289:
+ case 0x10ec0300:
alc_update_coef_idx(codec, 0x10, 1<<9, 0);
break;
case 0x10ec0275:
@@ -2830,6 +2831,7 @@ enum {
ALC269_TYPE_ALC215,
ALC269_TYPE_ALC225,
ALC269_TYPE_ALC294,
+ ALC269_TYPE_ALC300,
ALC269_TYPE_ALC700,
};
@@ -2864,6 +2866,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
case ALC269_TYPE_ALC215:
case ALC269_TYPE_ALC225:
case ALC269_TYPE_ALC294:
+ case ALC269_TYPE_ALC300:
case ALC269_TYPE_ALC700:
ssids = alc269_ssids;
break;
@@ -4985,9 +4988,18 @@ 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 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) {
@@ -5358,6 +5370,16 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec,
spec->gen.preferred_dacs = preferred_pairs;
}
+/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */
+static void alc285_fixup_invalidate_dacs(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action != HDA_FIXUP_ACT_PRE_PROBE)
+ return;
+
+ snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
/* for hda_fixup_thinkpad_acpi() */
#include "thinkpad_helper.c"
@@ -5492,6 +5514,9 @@ enum {
ALC255_FIXUP_DELL_HEADSET_MIC,
ALC295_FIXUP_HP_X360,
ALC221_FIXUP_HP_HEADSET_MIC,
+ ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
+ ALC295_FIXUP_HP_AUTO_MUTE,
+ ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5656,6 +5681,8 @@ static const struct hda_fixup alc269_fixups[] = {
[ALC269_FIXUP_HP_MUTE_LED_MIC3] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc269_fixup_hp_mute_led_mic3,
+ .chained = true,
+ .chain_id = ALC295_FIXUP_HP_AUTO_MUTE
},
[ALC269_FIXUP_HP_GPIO_LED] = {
.type = HDA_FIXUP_FUNC,
@@ -6359,6 +6386,23 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MIC
},
+ [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_invalidate_dacs,
+ },
+ [ALC295_FIXUP_HP_AUTO_MUTE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_auto_mute_via_amp,
+ },
+ [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x01a1913c }, /* 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[] = {
@@ -6373,7 +6417,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
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, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
+ SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -7032,6 +7080,15 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x12, 0x90a60130},
{0x19, 0x03a11020},
{0x21, 0x0321101f}),
+ SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
+ {0x12, 0x90a60130},
+ {0x14, 0x90170110},
+ {0x19, 0x04a11040},
+ {0x21, 0x04211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
+ {0x12, 0x90a60130},
+ {0x17, 0x90170110},
+ {0x21, 0x02211020}),
SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x12, 0x90a60120},
{0x14, 0x90170110},
@@ -7167,6 +7224,37 @@ static void alc269_fill_coef(struct hda_codec *codec)
alc_update_coef_idx(codec, 0x4, 0, 1<<11);
}
+static void alc294_hp_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+ int i, val;
+
+ if (!hp_pin)
+ return;
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+ msleep(100);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+ alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */
+ alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */
+
+ /* Wait for depop procedure finish */
+ val = alc_read_coefex_idx(codec, 0x58, 0x01);
+ for (i = 0; i < 20 && val & 0x0080; i++) {
+ msleep(50);
+ val = alc_read_coefex_idx(codec, 0x58, 0x01);
+ }
+ /* Set HP depop to auto mode */
+ alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b);
+ msleep(50);
+}
+
/*
*/
static int patch_alc269(struct hda_codec *codec)
@@ -7292,6 +7380,11 @@ static int patch_alc269(struct hda_codec *codec)
spec->codec_variant = ALC269_TYPE_ALC294;
spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */
alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */
+ alc294_hp_init(codec);
+ break;
+ case 0x10ec0300:
+ spec->codec_variant = ALC269_TYPE_ALC300;
+ spec->gen.mixer_nid = 0; /* no loopback on ALC300 */
break;
case 0x10ec0700:
case 0x10ec0701:
@@ -7299,6 +7392,7 @@ static int patch_alc269(struct hda_codec *codec)
spec->codec_variant = ALC269_TYPE_ALC700;
spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */
alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */
+ alc294_hp_init(codec);
break;
}
@@ -8403,6 +8497,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269),
HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 4e9854889a95..e63d6e33df48 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
*/
snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE,
AC_PWRST_D3);
- err = snd_hdac_display_power(bus, false);
- if (err < 0) {
- dev_err(dev, "Cannot turn on display power on i915\n");
- return err;
- }
hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev));
if (!hlink) {
@@ -2201,7 +2196,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
snd_hdac_ext_bus_link_put(bus, hlink);
- return 0;
+ err = snd_hdac_display_power(bus, false);
+ if (err < 0)
+ dev_err(dev, "Cannot turn off display power on i915\n");
+
+ return err;
}
static int hdac_hdmi_runtime_resume(struct device *dev)
diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h
index 2c6ba55bf394..bb3f0c42a1cd 100644
--- a/sound/soc/codecs/pcm186x.h
+++ b/sound/soc/codecs/pcm186x.h
@@ -139,7 +139,7 @@ enum pcm186x_type {
#define PCM186X_MAX_REGISTER PCM186X_CURR_TRIM_CTRL
/* PCM186X_PAGE */
-#define PCM186X_RESET 0xff
+#define PCM186X_RESET 0xfe
/* PCM186X_ADCX_INPUT_SEL_X */
#define PCM186X_ADC_INPUT_SEL_POL BIT(7)
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
index 494d9d662be8..771b46e1974b 100644
--- a/sound/soc/codecs/pcm3060.c
+++ b/sound/soc/codecs/pcm3060.c
@@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
};
static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
- SND_SOC_DAPM_OUTPUT("OUTL+"),
- SND_SOC_DAPM_OUTPUT("OUTR+"),
- SND_SOC_DAPM_OUTPUT("OUTL-"),
- SND_SOC_DAPM_OUTPUT("OUTR-"),
+ SND_SOC_DAPM_OUTPUT("OUTL"),
+ SND_SOC_DAPM_OUTPUT("OUTR"),
SND_SOC_DAPM_INPUT("INL"),
SND_SOC_DAPM_INPUT("INR"),
};
static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
- { "OUTL+", NULL, "Playback" },
- { "OUTR+", NULL, "Playback" },
- { "OUTL-", NULL, "Playback" },
- { "OUTR-", NULL, "Playback" },
+ { "OUTL", NULL, "Playback" },
+ { "OUTR", NULL, "Playback" },
{ "Capture", NULL, "INL" },
{ "Capture", NULL, "INR" },
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index a53dc174bbf0..66501b8dc46f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -765,38 +765,41 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem,
static void wm_adsp2_show_fw_status(struct wm_adsp *dsp)
{
- u16 scratch[4];
+ unsigned int scratch[4];
+ unsigned int addr = dsp->base + ADSP2_SCRATCH0;
+ unsigned int i;
int ret;
- ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0,
- scratch, sizeof(scratch));
- if (ret) {
- adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret);
- return;
+ for (i = 0; i < ARRAY_SIZE(scratch); ++i) {
+ ret = regmap_read(dsp->regmap, addr + i, &scratch[i]);
+ if (ret) {
+ adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret);
+ return;
+ }
}
adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n",
- be16_to_cpu(scratch[0]),
- be16_to_cpu(scratch[1]),
- be16_to_cpu(scratch[2]),
- be16_to_cpu(scratch[3]));
+ scratch[0], scratch[1], scratch[2], scratch[3]);
}
static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp)
{
- u32 scratch[2];
+ unsigned int scratch[2];
int ret;
- ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1,
- scratch, sizeof(scratch));
-
+ ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1,
+ &scratch[0]);
if (ret) {
- adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret);
+ adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret);
return;
}
- scratch[0] = be32_to_cpu(scratch[0]);
- scratch[1] = be32_to_cpu(scratch[1]);
+ ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3,
+ &scratch[1]);
+ if (ret) {
+ adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret);
+ return;
+ }
adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n",
scratch[0] & 0xFFFF,
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 0caa1f4eb94d..18e717703685 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -101,22 +101,42 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
codec, then enable this option by saying Y or m. This is a
recommended option
-config SND_SOC_INTEL_SKYLAKE_SSP_CLK
- tristate
-
config SND_SOC_INTEL_SKYLAKE
tristate "SKL/BXT/KBL/GLK/CNL... Platforms"
depends on PCI && ACPI
+ select SND_SOC_INTEL_SKYLAKE_COMMON
+ help
+ If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
+ GeminiLake or CannonLake platform with the DSP enabled in the BIOS
+ then enable this option by saying Y or m.
+
+if SND_SOC_INTEL_SKYLAKE
+
+config SND_SOC_INTEL_SKYLAKE_SSP_CLK
+ tristate
+
+config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+ bool "HDAudio codec support"
+ help
+ If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
+ GeminiLake or CannonLake platform with an HDaudio codec
+ then enable this option by saying Y
+
+config SND_SOC_INTEL_SKYLAKE_COMMON
+ tristate
select SND_HDA_EXT_CORE
select SND_HDA_DSP_LOADER
select SND_SOC_TOPOLOGY
select SND_SOC_INTEL_SST
+ select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
select SND_SOC_ACPI_INTEL_MATCH
help
If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
GeminiLake or CannonLake platform with the DSP enabled in the BIOS
then enable this option by saying Y or m.
+endif ## SND_SOC_INTEL_SKYLAKE
+
config SND_SOC_ACPI_INTEL_MATCH
tristate
select SND_SOC_ACPI if ACPI
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 73ca1350aa31..b177db2a0dbb 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -293,16 +293,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH
Say Y if you have such a device.
If unsure select "N".
-config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
- tristate "SKL/KBL/BXT/APL with HDA Codecs"
- select SND_SOC_HDAC_HDMI
- select SND_SOC_HDAC_HDA
- help
- This adds support for ASoC machine driver for Intel platforms
- SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
- Say Y or m if you have such a device. This is a recommended option.
- If unsure select "N".
-
config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
tristate "GLK with RT5682 and MAX98357A in I2S Mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
@@ -319,4 +309,18 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
endif ## SND_SOC_INTEL_SKYLAKE
+if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+
+config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
+ tristate "SKL/KBL/BXT/APL with HDA Codecs"
+ select SND_SOC_HDAC_HDMI
+ # SND_SOC_HDAC_HDA is already selected
+ help
+ This adds support for ASoC machine driver for Intel platforms
+ SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
+ Say Y or m if you have such a device. This is a recommended option.
+ If unsure select "N".
+
+endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+
endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index db6976f4ddaa..9d9f6e41d81c 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -19,6 +19,7 @@
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
+#include <linux/dmi.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
@@ -35,6 +36,8 @@
#define CHT_PLAT_CLK_3_HZ 19200000
#define CHT_CODEC_DAI "HiFi"
+#define QUIRK_PMC_PLT_CLK_0 0x01
+
struct cht_mc_private {
struct clk *mclk;
struct snd_soc_jack jack;
@@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = {
.num_controls = ARRAY_SIZE(cht_mc_controls),
};
+static const struct dmi_system_id cht_max98090_quirk_table[] = {
+ {
+ /* Swanky model Chromebook (Toshiba Chromebook 2) */
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"),
+ },
+ .driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+ },
+ {}
+};
+
static int snd_cht_mc_probe(struct platform_device *pdev)
{
+ const struct dmi_system_id *dmi_id;
struct device *dev = &pdev->dev;
int ret_val = 0;
struct cht_mc_private *drv;
+ const char *mclk_name;
+ int quirks = 0;
+
+ dmi_id = dmi_first_match(cht_max98090_quirk_table);
+ if (dmi_id)
+ quirks = (unsigned long)dmi_id->driver_data;
drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
if (!drv)
@@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
snd_soc_card_cht.dev = &pdev->dev;
snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
- drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
+ if (quirks & QUIRK_PMC_PLT_CLK_0)
+ mclk_name = "pmc_plt_clk_0";
+ else
+ mclk_name = "pmc_plt_clk_3";
+
+ drv->mclk = devm_clk_get(&pdev->dev, mclk_name);
if (IS_ERR(drv->mclk)) {
dev_err(&pdev->dev,
- "Failed to get MCLK from pmc_plt_clk_3: %ld\n",
- PTR_ERR(drv->mclk));
+ "Failed to get MCLK from %s: %ld\n",
+ mclk_name, PTR_ERR(drv->mclk));
return PTR_ERR(drv->mclk);
}
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 29225623b4b4..7487f388e65d 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -37,7 +37,9 @@
#include "skl.h"
#include "skl-sst-dsp.h"
#include "skl-sst-ipc.h"
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
#include "../../../soc/codecs/hdac_hda.h"
+#endif
/*
* initialize the PCI registers
@@ -658,6 +660,8 @@ static void skl_clock_device_unregister(struct skl *skl)
platform_device_unregister(skl->clk_dev);
}
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
+
#define IDISP_INTEL_VENDOR_ID 0x80860000
/*
@@ -676,6 +680,8 @@ static void load_codec_module(struct hda_codec *codec)
#endif
}
+#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */
+
/*
* Probe the given codec address
*/
@@ -685,9 +691,11 @@ static int probe_codec(struct hdac_bus *bus, int addr)
(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
unsigned int res = -1;
struct skl *skl = bus_to_skl(bus);
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
struct hdac_hda_priv *hda_codec;
- struct hdac_device *hdev;
int err;
+#endif
+ struct hdac_device *hdev;
mutex_lock(&bus->cmd_mutex);
snd_hdac_bus_send_cmd(bus, cmd);
@@ -697,6 +705,7 @@ static int probe_codec(struct hdac_bus *bus, int addr)
return -EIO;
dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res);
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec),
GFP_KERNEL);
if (!hda_codec)
@@ -715,6 +724,13 @@ static int probe_codec(struct hdac_bus *bus, int addr)
load_codec_module(&hda_codec->codec);
}
return 0;
+#else
+ hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
+ if (!hdev)
+ return -ENOMEM;
+
+ return snd_hdac_ext_bus_device_init(bus, addr, hdev);
+#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */
}
/* Codec initialization */
@@ -815,6 +831,12 @@ static void skl_probe_work(struct work_struct *work)
}
}
+ /*
+ * we are done probing so decrement link counts
+ */
+ list_for_each_entry(hlink, &bus->hlink_list, list)
+ snd_hdac_ext_bus_link_put(bus, hlink);
+
if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
err = snd_hdac_display_power(bus, false);
if (err < 0) {
@@ -824,12 +846,6 @@ static void skl_probe_work(struct work_struct *work)
}
}
- /*
- * we are done probing so decrement link counts
- */
- list_for_each_entry(hlink, &bus->hlink_list, list)
- snd_hdac_ext_bus_link_put(bus, hlink);
-
/* configure PM */
pm_runtime_put_noidle(bus->dev);
pm_runtime_allow(bus->dev);
@@ -870,7 +886,7 @@ static int skl_create(struct pci_dev *pci,
hbus = skl_to_hbus(skl);
bus = skl_to_bus(skl);
-#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA)
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
ext_ops = snd_soc_hdac_hda_get_ops();
#endif
snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index d5ae9eb8c756..fed45b41f9d3 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -36,6 +36,8 @@
#include "../codecs/twl6040.h"
struct abe_twl6040 {
+ struct snd_soc_card card;
+ struct snd_soc_dai_link dai_links[2];
int jack_detection; /* board can detect jack events */
int mclk_freq; /* MCLK frequency speed for twl6040 */
};
@@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
ARRAY_SIZE(dmic_audio_map));
}
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
- {
- .name = "TWL6040",
- .stream_name = "TWL6040",
- .codec_dai_name = "twl6040-legacy",
- .codec_name = "twl6040-codec",
- .init = omap_abe_twl6040_init,
- .ops = &omap_abe_ops,
- },
- {
- .name = "DMIC",
- .stream_name = "DMIC Capture",
- .codec_dai_name = "dmic-hifi",
- .codec_name = "dmic-codec",
- .init = omap_abe_dmic_init,
- .ops = &omap_abe_dmic_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card omap_abe_card = {
- .owner = THIS_MODULE,
-
- .dapm_widgets = twl6040_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
static int omap_abe_probe(struct platform_device *pdev)
{
struct device_node *node = pdev->dev.of_node;
- struct snd_soc_card *card = &omap_abe_card;
+ struct snd_soc_card *card;
struct device_node *dai_node;
struct abe_twl6040 *priv;
int num_links = 0;
@@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev)
return -ENODEV;
}
- card->dev = &pdev->dev;
-
priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
if (priv == NULL)
return -ENOMEM;
+ card = &priv->card;
+ card->dev = &pdev->dev;
+ card->owner = THIS_MODULE;
+ card->dapm_widgets = twl6040_dapm_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
+ card->dapm_routes = audio_map;
+ card->num_dapm_routes = ARRAY_SIZE(audio_map);
+
if (snd_soc_of_parse_card_name(card, "ti,model")) {
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
@@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "McPDM node is not provided\n");
return -EINVAL;
}
- abe_twl6040_dai_links[0].cpu_of_node = dai_node;
- abe_twl6040_dai_links[0].platform_of_node = dai_node;
+
+ priv->dai_links[0].name = "DMIC";
+ priv->dai_links[0].stream_name = "TWL6040";
+ priv->dai_links[0].cpu_of_node = dai_node;
+ priv->dai_links[0].platform_of_node = dai_node;
+ priv->dai_links[0].codec_dai_name = "twl6040-legacy";
+ priv->dai_links[0].codec_name = "twl6040-codec";
+ priv->dai_links[0].init = omap_abe_twl6040_init;
+ priv->dai_links[0].ops = &omap_abe_ops;
dai_node = of_parse_phandle(node, "ti,dmic", 0);
if (dai_node) {
num_links = 2;
- abe_twl6040_dai_links[1].cpu_of_node = dai_node;
- abe_twl6040_dai_links[1].platform_of_node = dai_node;
+ priv->dai_links[1].name = "TWL6040";
+ priv->dai_links[1].stream_name = "DMIC Capture";
+ priv->dai_links[1].cpu_of_node = dai_node;
+ priv->dai_links[1].platform_of_node = dai_node;
+ priv->dai_links[1].codec_dai_name = "dmic-hifi";
+ priv->dai_links[1].codec_name = "dmic-codec";
+ priv->dai_links[1].init = omap_abe_dmic_init;
+ priv->dai_links[1].ops = &omap_abe_dmic_ops;
} else {
num_links = 1;
}
@@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev)
return -ENODEV;
}
- card->dai_link = abe_twl6040_dai_links;
+ card->dai_link = priv->dai_links;
card->num_links = num_links;
snd_soc_card_set_drvdata(card, priv);
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index fe966272bd0c..cba9645b6487 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -48,6 +48,8 @@ struct omap_dmic {
struct device *dev;
void __iomem *io_base;
struct clk *fclk;
+ struct pm_qos_request pm_qos_req;
+ int latency;
int fclk_freq;
int out_freq;
int clk_div;
@@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
mutex_lock(&dmic->mutex);
+ pm_qos_remove_request(&dmic->pm_qos_req);
+
if (!dai->active)
dmic->active = 0;
@@ -228,6 +232,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
/* packet size is threshold * channels */
dma_data = snd_soc_dai_get_dma_data(dai, substream);
dma_data->maxburst = dmic->threshold * channels;
+ dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC /
+ params_rate(params);
return 0;
}
@@ -238,6 +244,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
u32 ctrl;
+ if (pm_qos_request_active(&dmic->pm_qos_req))
+ pm_qos_update_request(&dmic->pm_qos_req, dmic->latency);
+
/* Configure uplink threshold */
omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d0ebb6b9bfac..2d6decbfc99e 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -308,9 +308,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
pkt_size = channels;
}
- latency = ((((buffer_size - pkt_size) / channels) * 1000)
- / (params->rate_num / params->rate_den));
-
+ latency = (buffer_size - pkt_size) / channels;
+ latency = latency * USEC_PER_SEC /
+ (params->rate_num / params->rate_den);
mcbsp->latency[substream->stream] = latency;
omap_mcbsp_set_threshold(substream, pkt_size);
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 4c1be36c2207..7d5bdc5a2890 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -54,6 +54,8 @@ struct omap_mcpdm {
unsigned long phys_base;
void __iomem *io_base;
int irq;
+ struct pm_qos_request pm_qos_req;
+ int latency[2];
struct mutex mutex;
@@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+ int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+ int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock(&mcpdm->mutex);
@@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
}
}
+ if (mcpdm->latency[stream2])
+ pm_qos_update_request(&mcpdm->pm_qos_req,
+ mcpdm->latency[stream2]);
+ else if (mcpdm->latency[stream1])
+ pm_qos_remove_request(&mcpdm->pm_qos_req);
+
+ mcpdm->latency[stream1] = 0;
+
mutex_unlock(&mcpdm->mutex);
}
@@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
int stream = substream->stream;
struct snd_dmaengine_dai_dma_data *dma_data;
u32 threshold;
- int channels;
+ int channels, latency;
int link_mask = 0;
channels = params_channels(params);
@@ -344,14 +357,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
dma_data->maxburst =
(MCPDM_DN_THRES_MAX - threshold) * channels;
+ latency = threshold;
} else {
/* If playback is not running assume a stereo stream to come */
if (!mcpdm->config[!stream].link_mask)
mcpdm->config[!stream].link_mask = (0x3 << 3);
dma_data->maxburst = threshold * channels;
+ latency = (MCPDM_DN_THRES_MAX - threshold);
}
+ /*
+ * The DMA must act to a DMA request within latency time (usec) to avoid
+ * under/overflow
+ */
+ mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params);
+
+ if (!mcpdm->latency[stream])
+ mcpdm->latency[stream] = 10;
+
/* Check if we need to restart McPDM with this stream */
if (mcpdm->config[stream].link_mask &&
mcpdm->config[stream].link_mask != link_mask)
@@ -366,6 +390,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+ struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req;
+ int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+ int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ int latency = mcpdm->latency[stream2];
+
+ /* Prevent omap hardware from hitting off between FIFO fills */
+ if (!latency || mcpdm->latency[stream1] < latency)
+ latency = mcpdm->latency[stream1];
+
+ if (pm_qos_request_active(pm_qos_req))
+ pm_qos_update_request(pm_qos_req, latency);
+ else if (latency)
+ pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency);
if (!omap_mcpdm_active(mcpdm)) {
omap_mcpdm_start(mcpdm);
@@ -427,6 +465,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
free_irq(mcpdm->irq, (void *)mcpdm);
pm_runtime_disable(mcpdm->dev);
+ if (pm_qos_request_active(&mcpdm->pm_qos_req))
+ pm_qos_remove_request(&mcpdm->pm_qos_req);
+
return 0;
}
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index eb1b9da05dd4..4715527054e5 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -13,6 +13,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
struct device_node *cpu = NULL;
struct device *dev = card->dev;
struct snd_soc_dai_link *link;
+ struct of_phandle_args args;
int ret, num_links;
ret = snd_soc_of_parse_card_name(card, "model");
@@ -47,12 +48,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
goto err;
}
- link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
- if (!link->cpu_of_node) {
+ ret = of_parse_phandle_with_args(cpu, "sound-dai",
+ "#sound-dai-cells", 0, &args);
+ if (ret) {
dev_err(card->dev, "error getting cpu phandle\n");
- ret = -EINVAL;
goto err;
}
+ link->cpu_of_node = args.np;
+ link->id = args.args[0];
ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
if (ret) {
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 60ff4a2d3577..8f6c8fc073a9 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -1112,204 +1112,204 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
}
static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
- SND_SOC_DAPM_AIF_OUT("HDMI_RX", "HDMI Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_RX", "Slimbus2 Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_RX", "Slimbus3 Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback",
+ SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_MI2S_RX", "Tertiary MI2S Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_MI2S_TX", "Tertiary MI2S Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX", "Secondary MI2S Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_MI2S_TX", "Secondary MI2S Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX_SD1",
+ SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1",
"Secondary MI2S Playback SD1",
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRI_MI2S_RX", "Primary MI2S Playback",
+ SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRI_MI2S_TX", "Primary MI2S Capture",
+ SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_0", "Primary TDM0 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_1", "Primary TDM1 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_2", "Primary TDM2 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_3", "Primary TDM3 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_4", "Primary TDM4 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_5", "Primary TDM5 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_6", "Primary TDM6 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_7", "Primary TDM7 Playback",
+ SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_0", "Primary TDM0 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_1", "Primary TDM1 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_2", "Primary TDM2 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_3", "Primary TDM3 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_4", "Primary TDM4 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_5", "Primary TDM5 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_6", "Primary TDM6 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_7", "Primary TDM7 Capture",
+ SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_0", "Secondary TDM0 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_1", "Secondary TDM1 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_2", "Secondary TDM2 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_3", "Secondary TDM3 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_4", "Secondary TDM4 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_5", "Secondary TDM5 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_6", "Secondary TDM6 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_7", "Secondary TDM7 Playback",
+ SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_0", "Secondary TDM0 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_1", "Secondary TDM1 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_2", "Secondary TDM2 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_3", "Secondary TDM3 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_4", "Secondary TDM4 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_5", "Secondary TDM5 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_6", "Secondary TDM6 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_7", "Secondary TDM7 Capture",
+ SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_0", "Tertiary TDM0 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_1", "Tertiary TDM1 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_2", "Tertiary TDM2 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_3", "Tertiary TDM3 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_4", "Tertiary TDM4 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_5", "Tertiary TDM5 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_6", "Tertiary TDM6 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_7", "Tertiary TDM7 Playback",
+ SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_0", "Tertiary TDM0 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_1", "Tertiary TDM1 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_2", "Tertiary TDM2 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_3", "Tertiary TDM3 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_4", "Tertiary TDM4 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_5", "Tertiary TDM5 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_6", "Tertiary TDM6 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_7", "Tertiary TDM7 Capture",
+ SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_0", "Quaternary TDM0 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_1", "Quaternary TDM1 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_2", "Quaternary TDM2 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_3", "Quaternary TDM3 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_4", "Quaternary TDM4 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_5", "Quaternary TDM5 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_6", "Quaternary TDM6 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_7", "Quaternary TDM7 Playback",
+ SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_0", "Quaternary TDM0 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_1", "Quaternary TDM1 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_2", "Quaternary TDM2 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_3", "Quaternary TDM3 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_4", "Quaternary TDM4 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_5", "Quaternary TDM5 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_6", "Quaternary TDM6 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_7", "Quaternary TDM7 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_0", "Quinary TDM0 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_1", "Quinary TDM1 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_2", "Quinary TDM2 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_3", "Quinary TDM3 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_4", "Quinary TDM4 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_5", "Quinary TDM5 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_6", "Quinary TDM6 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_7", "Quinary TDM7 Playback",
+ SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_0", "Quinary TDM0 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_1", "Quinary TDM1 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_2", "Quinary TDM2 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_3", "Quinary TDM3 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_4", "Quinary TDM4 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_5", "Quinary TDM5 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_6", "Quinary TDM6 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_7", "Quinary TDM7 Capture",
+ SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL,
0, 0, 0, 0),
};
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 000775b4bba8..829b5e987b2a 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -49,14 +49,14 @@
#define AFE_PORT_I2S_SD1 0x2
#define AFE_PORT_I2S_SD2 0x3
#define AFE_PORT_I2S_SD3 0x4
-#define AFE_PORT_I2S_SD0_MASK BIT(0x1)
-#define AFE_PORT_I2S_SD1_MASK BIT(0x2)
-#define AFE_PORT_I2S_SD2_MASK BIT(0x3)
-#define AFE_PORT_I2S_SD3_MASK BIT(0x4)
-#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1)
-#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3)
-#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1)
-#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1)
+#define AFE_PORT_I2S_SD0_MASK BIT(0x0)
+#define AFE_PORT_I2S_SD1_MASK BIT(0x1)
+#define AFE_PORT_I2S_SD2_MASK BIT(0x2)
+#define AFE_PORT_I2S_SD3_MASK BIT(0x3)
+#define AFE_PORT_I2S_SD0_1_MASK GENMASK(1, 0)
+#define AFE_PORT_I2S_SD2_3_MASK GENMASK(3, 2)
+#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(2, 0)
+#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(3, 0)
#define AFE_PORT_I2S_QUAD01 0x5
#define AFE_PORT_I2S_QUAD23 0x6
#define AFE_PORT_I2S_6CHS 0x7
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index a16c71c03058..86115de5c1b2 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -122,7 +122,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
.rate_max = 48000, \
}, \
.name = "MultiMedia"#num, \
- .probe = fe_dai_probe, \
.id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
}
@@ -511,38 +510,6 @@ static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
}
}
-static const struct snd_soc_dapm_route afe_pcm_routes[] = {
- {"MM_DL1", NULL, "MultiMedia1 Playback" },
- {"MM_DL2", NULL, "MultiMedia2 Playback" },
- {"MM_DL3", NULL, "MultiMedia3 Playback" },
- {"MM_DL4", NULL, "MultiMedia4 Playback" },
- {"MM_DL5", NULL, "MultiMedia5 Playback" },
- {"MM_DL6", NULL, "MultiMedia6 Playback" },
- {"MM_DL7", NULL, "MultiMedia7 Playback" },
- {"MM_DL7", NULL, "MultiMedia8 Playback" },
- {"MultiMedia1 Capture", NULL, "MM_UL1"},
- {"MultiMedia2 Capture", NULL, "MM_UL2"},
- {"MultiMedia3 Capture", NULL, "MM_UL3"},
- {"MultiMedia4 Capture", NULL, "MM_UL4"},
- {"MultiMedia5 Capture", NULL, "MM_UL5"},
- {"MultiMedia6 Capture", NULL, "MM_UL6"},
- {"MultiMedia7 Capture", NULL, "MM_UL7"},
- {"MultiMedia8 Capture", NULL, "MM_UL8"},
-
-};
-
-static int fe_dai_probe(struct snd_soc_dai *dai)
-{
- struct snd_soc_dapm_context *dapm;
-
- dapm = snd_soc_component_get_dapm(dai->component);
- snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
- ARRAY_SIZE(afe_pcm_routes));
-
- return 0;
-}
-
-
static const struct snd_soc_component_driver q6asm_fe_dai_component = {
.name = DRV_NAME,
.ops = &q6asm_dai_ops,
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index c6b51571be94..d61b8404f7da 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -909,6 +909,25 @@ static const struct snd_soc_dapm_route intercon[] = {
{"MM_UL6", NULL, "MultiMedia6 Mixer"},
{"MM_UL7", NULL, "MultiMedia7 Mixer"},
{"MM_UL8", NULL, "MultiMedia8 Mixer"},
+
+ {"MM_DL1", NULL, "MultiMedia1 Playback" },
+ {"MM_DL2", NULL, "MultiMedia2 Playback" },
+ {"MM_DL3", NULL, "MultiMedia3 Playback" },
+ {"MM_DL4", NULL, "MultiMedia4 Playback" },
+ {"MM_DL5", NULL, "MultiMedia5 Playback" },
+ {"MM_DL6", NULL, "MultiMedia6 Playback" },
+ {"MM_DL7", NULL, "MultiMedia7 Playback" },
+ {"MM_DL8", NULL, "MultiMedia8 Playback" },
+
+ {"MultiMedia1 Capture", NULL, "MM_UL1"},
+ {"MultiMedia2 Capture", NULL, "MM_UL2"},
+ {"MultiMedia3 Capture", NULL, "MM_UL3"},
+ {"MultiMedia4 Capture", NULL, "MM_UL4"},
+ {"MultiMedia5 Capture", NULL, "MM_UL5"},
+ {"MultiMedia6 Capture", NULL, "MM_UL6"},
+ {"MultiMedia7 Capture", NULL, "MM_UL7"},
+ {"MultiMedia8 Capture", NULL, "MM_UL8"},
+
};
static int routing_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
index 9e7b5fa4cf59..4ac78d7a4b2d 100644
--- a/sound/soc/rockchip/rockchip_pcm.c
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -33,6 +33,7 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = {
static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = {
.pcm_hardware = &snd_rockchip_hardware,
+ .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
.prealloc_buffer_size = 32 * 1024,
};
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fcb4df23248c..6ec78f3096dd 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -306,7 +306,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
if (rsnd_ssi_is_multi_slave(mod, io))
return 0;
- if (ssi->rate) {
+ if (ssi->usrcnt > 1) {
if (ssi->rate != rate) {
dev_err(dev, "SSI parent/child should use same rate\n");
return -EINVAL;
diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c
index b8e72b52db30..4fb29f0e561e 100644
--- a/sound/soc/soc-acpi.c
+++ b/sound/soc/soc-acpi.c
@@ -10,11 +10,17 @@ struct snd_soc_acpi_mach *
snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines)
{
struct snd_soc_acpi_mach *mach;
+ struct snd_soc_acpi_mach *mach_alt;
for (mach = machines; mach->id[0]; mach++) {
if (acpi_dev_present(mach->id, NULL, -1)) {
- if (mach->machine_quirk)
- mach = mach->machine_quirk(mach);
+ if (mach->machine_quirk) {
+ mach_alt = mach->machine_quirk(mach);
+ if (!mach_alt)
+ continue; /* not full match, ignore */
+ mach = mach_alt;
+ }
+
return mach;
}
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6ddcf12bc030..b29d0f65611e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2131,6 +2131,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
card->instantiated = 1;
+ dapm_mark_endpoints_dirty(card);
snd_soc_dapm_sync(&card->dapm);
mutex_unlock(&card->mutex);
mutex_unlock(&client_mutex);
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index ea05cc91aa05..211589b0b2ef 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -390,7 +390,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai)
char *mclk_name, *p, *s = (char *)pname;
int ret, i = 0;
- mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL);
+ mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL);
if (!mclk)
return -ENOMEM;
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
index 66aad0d3f9c7..8134c3c94229 100644
--- a/sound/soc/sunxi/Kconfig
+++ b/sound/soc/sunxi/Kconfig
@@ -31,7 +31,7 @@ config SND_SUN8I_CODEC_ANALOG
config SND_SUN50I_CODEC_ANALOG
tristate "Allwinner sun50i Codec Analog Controls Support"
depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST
- select SND_SUNXI_ADDA_PR_REGMAP
+ select SND_SUN8I_ADDA_PR_REGMAP
help
Say Y or M if you want to add support for the analog controls for
the codec embedded in Allwinner A64 SoC.
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 522a72fde78d..92c5de026c43 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -481,7 +481,11 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = {
{ "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch",
"AIF1 Slot 0 Right"},
- /* ADC routes */
+ /* ADC Routes */
+ { "AIF1 Slot 0 Right ADC", NULL, "ADC" },
+ { "AIF1 Slot 0 Left ADC", NULL, "ADC" },
+
+ /* ADC Mixer Routes */
{ "Left Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch",
"AIF1 Slot 0 Left ADC" },
{ "Right Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch",
@@ -605,16 +609,10 @@ err_pm_disable:
static int sun8i_codec_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- struct sun8i_codec *scodec = snd_soc_card_get_drvdata(card);
-
pm_runtime_disable(&pdev->dev);
if (!pm_runtime_status_suspended(&pdev->dev))
sun8i_codec_runtime_suspend(&pdev->dev);
- clk_disable_unprepare(scodec->clk_module);
- clk_disable_unprepare(scodec->clk_bus);
-
return 0;
}
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 2bfe4e80a6b9..a105947eaf55 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -682,9 +682,12 @@ static int usb_audio_probe(struct usb_interface *intf,
__error:
if (chip) {
+ /* chip->active is inside the chip->card object,
+ * decrement before memory is possibly returned.
+ */
+ atomic_dec(&chip->active);
if (!chip->num_interfaces)
snd_card_free(chip->card);
- atomic_dec(&chip->active);
}
mutex_unlock(&register_mutex);
return err;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 849953e5775c..37fc0447c071 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
.ifnum = QUIRK_NO_INTERFACE
}
},
+/* Dell WD19 Dock */
+{
+ USB_DEVICE(0x0bda, 0x402e),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Dell",
+ .product_name = "WD19 Dock",
+ .profile_name = "Dell-WD15-Dock",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
#undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 3d0f09108c98..96340f23f86d 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1492,6 +1492,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;
+ case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */
case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */
case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */