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-rw-r--r--sound/isa/es18xx.c10
-rw-r--r--sound/pci/hda/hda_controller.c34
-rw-r--r--sound/pci/hda/hda_intel.c9
-rw-r--r--sound/pci/hda/hda_priv.h1
-rw-r--r--sound/pci/hda/patch_hdmi.c6
-rw-r--r--sound/pci/hda/patch_realtek.c10
-rw-r--r--sound/soc/codecs/alc5623.c2
-rw-r--r--sound/soc/codecs/cs42l52.c6
-rw-r--r--sound/soc/codecs/cs42l73.c6
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c9
-rw-r--r--sound/soc/codecs/wm8962.c15
-rw-r--r--sound/soc/codecs/wm8962.h4
-rw-r--r--sound/soc/fsl/fsl_esai.c22
-rw-r--r--sound/soc/fsl/fsl_spdif.h4
-rw-r--r--sound/soc/fsl/imx-audmux.c2
-rw-r--r--sound/soc/jz4740/Makefile2
-rw-r--r--sound/soc/sh/rcar/core.c5
-rw-r--r--sound/soc/sh/rcar/src.c4
-rw-r--r--sound/soc/sh/rcar/ssi.c4
-rw-r--r--sound/soc/soc-dapm.c15
-rw-r--r--sound/soc/soc-pcm.c2
-rw-r--r--sound/usb/card.c12
-rw-r--r--sound/usb/card.h1
-rw-r--r--sound/usb/endpoint.c15
-rw-r--r--sound/usb/pcm.c5
-rw-r--r--sound/usb/usbaudio.h1
27 files changed, 140 insertions, 68 deletions
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index 1c16830af3d8..6faaac60161a 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -520,7 +520,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip,
snd_es18xx_mixer_write(chip, 0x78, 0x93);
#ifdef AVOID_POPS
/* Avoid pops */
- udelay(100000);
+ mdelay(100);
if (chip->caps & ES18XX_PCM2)
/* Restore Audio 2 volume */
snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol);
@@ -537,7 +537,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip,
/* Stop DMA */
snd_es18xx_mixer_write(chip, 0x78, 0x00);
#ifdef AVOID_POPS
- udelay(25000);
+ mdelay(25);
if (chip->caps & ES18XX_PCM2)
/* Set Audio 2 volume to 0 */
snd_es18xx_mixer_write(chip, 0x7C, 0);
@@ -596,7 +596,7 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream)
snd_es18xx_write(chip, 0xA5, count >> 8);
#ifdef AVOID_POPS
- udelay(100000);
+ mdelay(100);
#endif
/* Set format */
@@ -691,7 +691,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip,
snd_es18xx_write(chip, 0xB8, 0x05);
#ifdef AVOID_POPS
/* Avoid pops */
- udelay(100000);
+ mdelay(100);
/* Enable Audio 1 */
snd_es18xx_dsp_command(chip, 0xD1);
#endif
@@ -705,7 +705,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip,
snd_es18xx_write(chip, 0xB8, 0x00);
#ifdef AVOID_POPS
/* Avoid pops */
- udelay(25000);
+ mdelay(25);
/* Disable Audio 1 */
snd_es18xx_dsp_command(chip, 0xD3);
#endif
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 248b90abb882..480bbddbd801 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1059,24 +1059,26 @@ static void azx_init_cmd_io(struct azx *chip)
/* reset the corb hw read pointer */
azx_writew(chip, CORBRP, ICH6_CORBRP_RST);
- for (timeout = 1000; timeout > 0; timeout--) {
- if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST)
- break;
- udelay(1);
- }
- if (timeout <= 0)
- dev_err(chip->card->dev, "CORB reset timeout#1, CORBRP = %d\n",
- azx_readw(chip, CORBRP));
+ if (!(chip->driver_caps & AZX_DCAPS_CORBRP_SELF_CLEAR)) {
+ for (timeout = 1000; timeout > 0; timeout--) {
+ if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST)
+ break;
+ udelay(1);
+ }
+ if (timeout <= 0)
+ dev_err(chip->card->dev, "CORB reset timeout#1, CORBRP = %d\n",
+ azx_readw(chip, CORBRP));
- azx_writew(chip, CORBRP, 0);
- for (timeout = 1000; timeout > 0; timeout--) {
- if (azx_readw(chip, CORBRP) == 0)
- break;
- udelay(1);
+ azx_writew(chip, CORBRP, 0);
+ for (timeout = 1000; timeout > 0; timeout--) {
+ if (azx_readw(chip, CORBRP) == 0)
+ break;
+ udelay(1);
+ }
+ if (timeout <= 0)
+ dev_err(chip->card->dev, "CORB reset timeout#2, CORBRP = %d\n",
+ azx_readw(chip, CORBRP));
}
- if (timeout <= 0)
- dev_err(chip->card->dev, "CORB reset timeout#2, CORBRP = %d\n",
- azx_readw(chip, CORBRP));
/* enable corb dma */
azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d6bca62ef387..2c54629d62d1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -249,7 +249,8 @@ enum {
/* quirks for Nvidia */
#define AZX_DCAPS_PRESET_NVIDIA \
(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
- AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT)
+ AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT |\
+ AZX_DCAPS_CORBRP_SELF_CLEAR)
#define AZX_DCAPS_PRESET_CTHDA \
(AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY)
@@ -1366,6 +1367,12 @@ static int azx_first_init(struct azx *chip)
/* initialize streams */
azx_init_stream(chip);
+ /* workaround for Broadwell HDMI: the first stream is broken,
+ * so mask it by keeping it as if opened
+ */
+ if (pci->vendor == 0x8086 && pci->device == 0x160c)
+ chip->azx_dev[0].opened = 1;
+
/* initialize chip */
azx_init_pci(chip);
azx_init_chip(chip, (probe_only[dev] & 2) == 0);
diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h
index ba38b819f984..4a7cb01fa912 100644
--- a/sound/pci/hda/hda_priv.h
+++ b/sound/pci/hda/hda_priv.h
@@ -189,6 +189,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */
#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */
#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */
+#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */
/* position fix mode */
enum {
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 0cb5b89cd0c8..b4218a19df22 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1127,8 +1127,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
AMP_OUT_UNMUTE);
eld = &per_pin->sink_eld;
- if (!eld->monitor_present)
+ if (!eld->monitor_present) {
+ hdmi_set_channel_count(codec, per_pin->cvt_nid, channels);
return;
+ }
if (!non_pcm && per_pin->chmap_set)
ca = hdmi_manual_channel_allocation(channels, per_pin->chmap);
@@ -3330,6 +3332,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi },
{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi },
{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
+{ .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi },
{ .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch },
{ .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
{ .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
@@ -3385,6 +3388,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0044");
MODULE_ALIAS("snd-hda-codec-id:10de0051");
MODULE_ALIAS("snd-hda-codec-id:10de0060");
MODULE_ALIAS("snd-hda-codec-id:10de0067");
+MODULE_ALIAS("snd-hda-codec-id:10de0071");
MODULE_ALIAS("snd-hda-codec-id:10de8001");
MODULE_ALIAS("snd-hda-codec-id:11069f80");
MODULE_ALIAS("snd-hda-codec-id:11069f81");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 14ae979a92ea..49e884fb3e5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4616,11 +4616,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0657, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x065c, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0674, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x067e, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0680, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0684, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -4912,6 +4918,7 @@ static int patch_alc269(struct hda_codec *codec)
spec->codec_variant = ALC269_TYPE_ALC285;
break;
case 0x10ec0286:
+ case 0x10ec0288:
spec->codec_variant = ALC269_TYPE_ALC286;
break;
case 0x10ec0255:
@@ -5539,6 +5546,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE),
SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_AUTO_MUTE),
+ SND_PCI_QUIRK(0x1028, 0x0696, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A),
SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
@@ -5781,6 +5790,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
{ .id = 0x10ec0285, .name = "ALC285", .patch = patch_alc269 },
{ .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 },
+ { .id = 0x10ec0288, .name = "ALC288", .patch = patch_alc269 },
{ .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
{ .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
{ .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 },
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index f500905e9373..2acf82f4a08a 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -1018,13 +1018,13 @@ static int alc5623_i2c_probe(struct i2c_client *client,
dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret);
return ret;
}
- vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2);
if (ret < 0) {
dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret);
return ret;
}
+ vid2 >>= 8;
if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
dev_err(&client->dev, "unknown or wrong codec\n");
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 460d35547a68..2213a037c893 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -1229,8 +1229,10 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
}
if (cs42l52->pdata.reset_gpio) {
- ret = gpio_request_one(cs42l52->pdata.reset_gpio,
- GPIOF_OUT_INIT_HIGH, "CS42L52 /RST");
+ ret = devm_gpio_request_one(&i2c_client->dev,
+ cs42l52->pdata.reset_gpio,
+ GPIOF_OUT_INIT_HIGH,
+ "CS42L52 /RST");
if (ret < 0) {
dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n",
cs42l52->pdata.reset_gpio, ret);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 0ee60a19a263..ae3717992d56 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1443,8 +1443,10 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
i2c_set_clientdata(i2c_client, cs42l73);
if (cs42l73->pdata.reset_gpio) {
- ret = gpio_request_one(cs42l73->pdata.reset_gpio,
- GPIOF_OUT_INIT_HIGH, "CS42L73 /RST");
+ ret = devm_gpio_request_one(&i2c_client->dev,
+ cs42l73->pdata.reset_gpio,
+ GPIOF_OUT_INIT_HIGH,
+ "CS42L73 /RST");
if (ret < 0) {
dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n",
cs42l73->pdata.reset_gpio, ret);
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index fa158cfe9b32..d1929de641e2 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -376,7 +376,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
reg = AIC31XX_ADCFLAG;
break;
default:
- dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
+ dev_err(w->codec->dev, "Unknown widget '%s' calling %s\n",
w->name, __func__);
return -EINVAL;
}
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b1835103e9b4..d7349bc89ad3 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1399,7 +1399,6 @@ static int aic3x_probe(struct snd_soc_codec *codec)
}
aic3x_add_widgets(codec);
- list_add(&aic3x->list, &reset_list);
return 0;
@@ -1569,7 +1568,13 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_aic3x, &aic3x_dai, 1);
- return ret;
+
+ if (ret != 0)
+ goto err_gpio;
+
+ list_add(&aic3x->list, &reset_list);
+
+ return 0;
err_gpio:
if (gpio_is_valid(aic3x->gpio_reset) &&
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 5522d2566c67..ecd26dd2e442 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -154,6 +154,7 @@ static struct reg_default wm8962_reg[] = {
{ 40, 0x0000 }, /* R40 - SPKOUTL volume */
{ 41, 0x0000 }, /* R41 - SPKOUTR volume */
+ { 49, 0x0010 }, /* R49 - Class D Control 1 */
{ 51, 0x0003 }, /* R51 - Class D Control 2 */
{ 56, 0x0506 }, /* R56 - Clocking 4 */
@@ -795,7 +796,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg)
case WM8962_ALC2:
case WM8962_THERMAL_SHUTDOWN_STATUS:
case WM8962_ADDITIONAL_CONTROL_4:
- case WM8962_CLASS_D_CONTROL_1:
case WM8962_DC_SERVO_6:
case WM8962_INTERRUPT_STATUS_1:
case WM8962_INTERRUPT_STATUS_2:
@@ -2929,13 +2929,22 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
static int wm8962_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- int val;
+ int val, ret;
if (mute)
- val = WM8962_DAC_MUTE;
+ val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT;
else
val = 0;
+ /**
+ * The DAC mute bit is mirrored in two registers, update both to keep
+ * the register cache consistent.
+ */
+ ret = snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_1,
+ WM8962_DAC_MUTE_ALT, val);
+ if (ret < 0)
+ return ret;
+
return snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1,
WM8962_DAC_MUTE, val);
}
diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h
index a1a5d5294c19..910aafd09d21 100644
--- a/sound/soc/codecs/wm8962.h
+++ b/sound/soc/codecs/wm8962.h
@@ -1954,6 +1954,10 @@
#define WM8962_SPKOUTL_ENA_MASK 0x0040 /* SPKOUTL_ENA */
#define WM8962_SPKOUTL_ENA_SHIFT 6 /* SPKOUTL_ENA */
#define WM8962_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */
+#define WM8962_DAC_MUTE_ALT 0x0010 /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_MASK 0x0010 /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_SHIFT 4 /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_WIDTH 1 /* DAC_MUTE */
#define WM8962_SPKOUTL_PGA_MUTE 0x0002 /* SPKOUTL_PGA_MUTE */
#define WM8962_SPKOUTL_PGA_MUTE_MASK 0x0002 /* SPKOUTL_PGA_MUTE */
#define WM8962_SPKOUTL_PGA_MUTE_SHIFT 1 /* SPKOUTL_PGA_MUTE */
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c8e5db1414d7..496ce2eb2f1f 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -258,10 +258,16 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
return -EINVAL;
}
- if (ratio == 1) {
+ /* Only EXTAL source can be output directly without using PSR and PM */
+ if (ratio == 1 && clksrc == esai_priv->extalclk) {
/* Bypass all the dividers if not being needed */
ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO;
goto out;
+ } else if (ratio < 2) {
+ /* The ratio should be no less than 2 if using other sources */
+ dev_err(dai->dev, "failed to derive required HCK%c rate\n",
+ tx ? 'T' : 'R');
+ return -EINVAL;
}
ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0);
@@ -307,7 +313,8 @@ static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
return -EINVAL;
}
- if (esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) {
+ /* The ratio should be contented by FP alone if bypassing PM and PSR */
+ if (!esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) {
dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n");
return -EINVAL;
}
@@ -454,12 +461,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
}
if (!dai->active) {
- /* Reset Port C */
- regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
- ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
- regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
- ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
-
/* Set synchronous mode */
regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR,
ESAI_SAICR_SYNC, esai_priv->synchronous ?
@@ -519,6 +520,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
+ /* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
+ ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
+ ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
return 0;
}
diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h
index b1266790d117..605a10b2112b 100644
--- a/sound/soc/fsl/fsl_spdif.h
+++ b/sound/soc/fsl/fsl_spdif.h
@@ -144,8 +144,8 @@ enum spdif_gainsel {
/* SPDIF Clock register */
#define STC_SYSCLK_DIV_OFFSET 11
-#define STC_SYSCLK_DIV_MASK (0x1ff << STC_TXCLK_SRC_OFFSET)
-#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK)
+#define STC_SYSCLK_DIV_MASK (0x1ff << STC_SYSCLK_DIV_OFFSET)
+#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_SYSCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK)
#define STC_TXCLK_SRC_OFFSET 8
#define STC_TXCLK_SRC_MASK (0x7 << STC_TXCLK_SRC_OFFSET)
#define STC_TXCLK_SRC_SET(x) ((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK)
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index ac869931d7f1..267717aa96c1 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -145,7 +145,7 @@ static const struct file_operations audmux_debugfs_fops = {
.llseek = default_llseek,
};
-static void __init audmux_debugfs_init(void)
+static void audmux_debugfs_init(void)
{
int i;
char buf[20];
diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile
index be873c1b0c20..d32c540555c4 100644
--- a/sound/soc/jz4740/Makefile
+++ b/sound/soc/jz4740/Makefile
@@ -1,10 +1,8 @@
#
# Jz4740 Platform Support
#
-snd-soc-jz4740-objs := jz4740-pcm.o
snd-soc-jz4740-i2s-objs := jz4740-i2s.o
-obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o
obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o
# Jz4740 Machine Support
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 215b668166be..89424470a1f3 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -197,13 +197,12 @@ static void rsnd_dma_complete(void *data)
* rsnd_dai_pointer_update() will be called twice,
* ant it will breaks io->byte_pos
*/
-
- rsnd_dai_pointer_update(io, io->byte_per_period);
-
if (dma->submit_loop)
rsnd_dma_continue(dma);
rsnd_unlock(priv, flags);
+
+ rsnd_dai_pointer_update(io, io->byte_per_period);
}
static void __rsnd_dma_start(struct rsnd_dma *dma)
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 6232b7d307aa..4d0720ed5a90 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -258,7 +258,7 @@ static int rsnd_src_init(struct rsnd_mod *mod,
{
struct rsnd_src *src = rsnd_mod_to_src(mod);
- clk_enable(src->clk);
+ clk_prepare_enable(src->clk);
return 0;
}
@@ -269,7 +269,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod,
{
struct rsnd_src *src = rsnd_mod_to_src(mod);
- clk_disable(src->clk);
+ clk_disable_unprepare(src->clk);
return 0;
}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 4b7e20603dd7..1d8387c25bd8 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -171,7 +171,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
u32 cr;
if (0 == ssi->usrcnt) {
- clk_enable(ssi->clk);
+ clk_prepare_enable(ssi->clk);
if (rsnd_dai_is_clk_master(rdai)) {
if (rsnd_ssi_clk_from_parent(ssi))
@@ -230,7 +230,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi,
rsnd_ssi_master_clk_stop(ssi);
}
- clk_disable(ssi->clk);
+ clk_disable_unprepare(ssi->clk);
}
dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod));
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c8a780d0d057..6d6ceee447d5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -254,7 +254,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
static void dapm_kcontrol_free(struct snd_kcontrol *kctl)
{
struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl);
- kfree(data->widget);
kfree(data->wlist);
kfree(data);
}
@@ -1613,8 +1612,11 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie)
"ASoC: Failed to turn on bias: %d\n", ret);
}
- /* Prepare for a STADDBY->ON or ON->STANDBY transition */
- if (d->bias_level != d->target_bias_level) {
+ /* Prepare for a transition to ON or away from ON */
+ if ((d->target_bias_level == SND_SOC_BIAS_ON &&
+ d->bias_level != SND_SOC_BIAS_ON) ||
+ (d->target_bias_level != SND_SOC_BIAS_ON &&
+ d->bias_level == SND_SOC_BIAS_ON)) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
if (ret != 0)
dev_err(d->dev,
@@ -3476,8 +3478,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
cpu_dai = rtd->cpu_dai;
codec_dai = rtd->codec_dai;
- /* dynamic FE links have no fixed DAI mapping */
- if (rtd->dai_link->dynamic)
+ /*
+ * dynamic FE links have no fixed DAI mapping.
+ * CODEC<->CODEC links have no direct connection.
+ */
+ if (rtd->dai_link->dynamic || rtd->dai_link->params)
continue;
/* there is no point in connecting BE DAI links with dummies */
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2cedf09f6d96..a391de058037 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1675,7 +1675,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
- if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
continue;
if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 893d5a1afc3c..c3b5b7dca1c3 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -651,7 +651,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip)
int err = -ENODEV;
down_read(&chip->shutdown_rwsem);
- if (chip->probing)
+ if (chip->probing && chip->in_pm)
err = 0;
else if (!chip->shutdown)
err = usb_autopm_get_interface(chip->pm_intf);
@@ -663,7 +663,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip)
void snd_usb_autosuspend(struct snd_usb_audio *chip)
{
down_read(&chip->shutdown_rwsem);
- if (!chip->shutdown && !chip->probing)
+ if (!chip->shutdown && !chip->probing && !chip->in_pm)
usb_autopm_put_interface(chip->pm_intf);
up_read(&chip->shutdown_rwsem);
}
@@ -695,8 +695,9 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
chip->autosuspended = 1;
}
- list_for_each_entry(mixer, &chip->mixer_list, list)
- snd_usb_mixer_suspend(mixer);
+ if (chip->num_suspended_intf == 1)
+ list_for_each_entry(mixer, &chip->mixer_list, list)
+ snd_usb_mixer_suspend(mixer);
return 0;
}
@@ -711,6 +712,8 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
return 0;
if (--chip->num_suspended_intf)
return 0;
+
+ chip->in_pm = 1;
/*
* ALSA leaves material resumption to user space
* we just notify and restart the mixers
@@ -726,6 +729,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
chip->autosuspended = 0;
err_out:
+ chip->in_pm = 0;
return err;
}
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 9867ab866857..97acb906acc2 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -92,6 +92,7 @@ struct snd_usb_endpoint {
unsigned int curframesize; /* current packet size in frames (for capture) */
unsigned int syncmaxsize; /* sync endpoint packet size */
unsigned int fill_max:1; /* fill max packet size always */
+ unsigned int udh01_fb_quirk:1; /* corrupted feedback data */
unsigned int datainterval; /* log_2 of data packet interval */
unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */
unsigned char silence_value;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index e70a87e0d9fe..289f582c9130 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -471,6 +471,10 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
ep->syncinterval = 3;
ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize);
+
+ if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ &&
+ ep->syncmaxsize == 4)
+ ep->udh01_fb_quirk = 1;
}
list_add_tail(&ep->list, &chip->ep_list);
@@ -1105,7 +1109,16 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
if (f == 0)
return;
- if (unlikely(ep->freqshift == INT_MIN)) {
+ if (unlikely(sender->udh01_fb_quirk)) {
+ /*
+ * The TEAC UD-H01 firmware sometimes changes the feedback value
+ * by +/- 0x1.0000.
+ */
+ if (f < ep->freqn - 0x8000)
+ f += 0x10000;
+ else if (f > ep->freqn + 0x8000)
+ f -= 0x10000;
+ } else if (unlikely(ep->freqshift == INT_MIN)) {
/*
* The first time we see a feedback value, determine its format
* by shifting it left or right until it matches the nominal
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 131336d40492..c62a1659106d 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1501,9 +1501,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
* The error should be lower than 2ms since the estimate relies
* on two reads of a counter updated every ms.
*/
- if (printk_ratelimit() &&
- abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
- dev_dbg(&subs->dev->dev,
+ if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
+ dev_dbg_ratelimited(&subs->dev->dev,
"delay: estimated %d, actual %d\n",
est_delay, subs->last_delay);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 25c4c7e217de..91d0380431b4 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -40,6 +40,7 @@ struct snd_usb_audio {
struct rw_semaphore shutdown_rwsem;
unsigned int shutdown:1;
unsigned int probing:1;
+ unsigned int in_pm:1;
unsigned int autosuspended:1;
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
generated by cgit v1.2.3-58-ga151 (git 2.39.0) at 2024-12-28 01:33:24 +0000