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-rw-r--r--sound/pci/hda/hda_generic.c47
-rw-r--r--sound/pci/hda/hda_generic.h3
-rw-r--r--sound/pci/hda/patch_analog.c10
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_hdmi.c5
-rw-r--r--sound/pci/hda/patch_realtek.c11
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c30
-rw-r--r--sound/soc/atmel/sam9x5_wm8731.c2
-rw-r--r--sound/soc/codecs/wm5110.c2
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/codecs/wm8962.c13
-rw-r--r--sound/soc/codecs/wm_adsp.c10
-rw-r--r--sound/soc/fsl/imx-wm8962.c2
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c38
-rw-r--r--sound/soc/soc-pcm.c5
-rw-r--r--sound/soc/tegra/tegra20_i2s.c6
-rw-r--r--sound/soc/tegra/tegra20_spdif.c10
-rw-r--r--sound/soc/tegra/tegra30_i2s.c6
-rw-r--r--sound/usb/mixer_quirks.c2
19 files changed, 167 insertions, 38 deletions
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index c4671d0..c7f6d1c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -474,6 +474,20 @@ static void invalidate_nid_path(struct hda_codec *codec, int idx)
memset(path, 0, sizeof(*path));
}
+/* return a DAC if paired to the given pin by codec driver */
+static hda_nid_t get_preferred_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ const hda_nid_t *list = spec->preferred_dacs;
+
+ if (!list)
+ return 0;
+ for (; *list; list += 2)
+ if (*list == pin)
+ return list[1];
+ return 0;
+}
+
/* look for an empty DAC slot */
static hda_nid_t look_for_dac(struct hda_codec *codec, hda_nid_t pin,
bool is_digital)
@@ -1192,7 +1206,14 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs,
continue;
}
- dacs[i] = look_for_dac(codec, pin, false);
+ dacs[i] = get_preferred_dac(codec, pin);
+ if (dacs[i]) {
+ if (is_dac_already_used(codec, dacs[i]))
+ badness += bad->shared_primary;
+ }
+
+ if (!dacs[i])
+ dacs[i] = look_for_dac(codec, pin, false);
if (!dacs[i] && !i) {
/* try to steal the DAC of surrounds for the front */
for (j = 1; j < num_outs; j++) {
@@ -4297,6 +4318,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
return AC_PWRST_D3;
}
+/* mute all aamix inputs initially; parse up to the first leaves */
+static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
+{
+ int i, nums;
+ const hda_nid_t *conn;
+ bool has_amp;
+
+ nums = snd_hda_get_conn_list(codec, mix, &conn);
+ has_amp = nid_has_mute(codec, mix, HDA_INPUT);
+ for (i = 0; i < nums; i++) {
+ if (has_amp)
+ snd_hda_codec_amp_stereo(codec, mix,
+ HDA_INPUT, i,
+ 0xff, HDA_AMP_MUTE);
+ else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
+ snd_hda_codec_amp_stereo(codec, conn[i],
+ HDA_OUTPUT, 0,
+ 0xff, HDA_AMP_MUTE);
+ }
+}
/*
* Parse the given BIOS configuration and set up the hda_gen_spec
@@ -4435,6 +4476,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
}
}
+ /* mute all aamix input initially */
+ if (spec->mixer_nid)
+ mute_all_mixer_nid(codec, spec->mixer_nid);
+
dig_only:
parse_digital(codec);
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 7e45cb4..0929a06 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -249,6 +249,9 @@ struct hda_gen_spec {
const struct badness_table *main_out_badness;
const struct badness_table *extra_out_badness;
+ /* preferred pin/DAC pairs; an array of paired NIDs */
+ const hda_nid_t *preferred_dacs;
+
/* loopback mixing mode */
bool aamix_mode;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index cac015b..699262a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -340,6 +340,14 @@ static int patch_ad1986a(struct hda_codec *codec)
{
int err;
struct ad198x_spec *spec;
+ static hda_nid_t preferred_pairs[] = {
+ 0x1a, 0x03,
+ 0x1b, 0x03,
+ 0x1c, 0x04,
+ 0x1d, 0x05,
+ 0x1e, 0x03,
+ 0
+ };
err = alloc_ad_spec(codec);
if (err < 0)
@@ -360,6 +368,8 @@ static int patch_ad1986a(struct hda_codec *codec)
* So, let's disable the shared stream.
*/
spec->gen.multiout.no_share_stream = 1;
+ /* give fixed DAC/pin pairs */
+ spec->gen.preferred_dacs = preferred_pairs;
/* AD1986A can't manage the dynamic pin on/off smoothly */
spec->gen.auto_mute_via_amp = 1;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 1f2717f..3fbf288 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
- SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index c4a66ef..f281c80 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2337,8 +2337,9 @@ static int simple_playback_build_controls(struct hda_codec *codec)
int err;
per_cvt = get_cvt(spec, 0);
- err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid,
- per_cvt->cvt_nid);
+ err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid,
+ per_cvt->cvt_nid,
+ HDA_PCM_TYPE_HDMI);
if (err < 0)
return err;
return simple_hdmi_build_jack(codec, 0);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c5ea483..34de5dc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3849,6 +3849,7 @@ enum {
ALC269_FIXUP_ASUS_X101,
ALC271_FIXUP_AMIC_MIC2,
ALC271_FIXUP_HP_GATE_MIC_JACK,
+ ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572,
ALC269_FIXUP_ACER_AC700,
ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
ALC269VB_FIXUP_ASUS_ZENBOOK,
@@ -4111,6 +4112,12 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC271_FIXUP_AMIC_MIC2,
},
+ [ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_limit_int_mic_boost,
+ .chained = true,
+ .chain_id = ALC271_FIXUP_HP_GATE_MIC_JACK,
+ },
[ALC269_FIXUP_ACER_AC700] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -4208,6 +4215,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+ SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
@@ -5034,8 +5042,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0628, "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_CHMAP),
SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP),
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 8697ced..1ead3c9 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
dma_params = ssc_p->dma_params[dir];
- ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error);
pr_debug("%s enabled SSC_SR=0x%08x\n",
@@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
return 0;
}
+static int atmel_ssc_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
+ struct atmel_pcm_dma_params *dma_params;
+ int dir;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = 0;
+ else
+ dir = 1;
+
+ dma_params = ssc_p->dma_params[dir];
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+ break;
+ default:
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+ break;
+ }
+
+ return 0;
+}
#ifdef CONFIG_PM
static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
@@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = {
.startup = atmel_ssc_startup,
.shutdown = atmel_ssc_shutdown,
.prepare = atmel_ssc_prepare,
+ .trigger = atmel_ssc_trigger,
.hw_params = atmel_ssc_hw_params,
.set_fmt = atmel_ssc_set_dai_fmt,
.set_clkdiv = atmel_ssc_set_dai_clkdiv,
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 1b37228..7d6a905 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -109,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
dai->stream_name = "WM8731 PCM";
dai->codec_dai_name = "wm8731-hifi";
dai->init = sam9x5_wm8731_init;
- dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM;
ret = snd_soc_of_parse_card_name(card, "atmel,model");
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 99b359e..0ab2dc2 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1012,7 +1012,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
{ "AEC Loopback", "HPOUT3L", "OUT3L" },
{ "AEC Loopback", "HPOUT3R", "OUT3R" },
{ "HPOUT3L", NULL, "OUT3L" },
- { "HPOUT3R", NULL, "OUT3L" },
+ { "HPOUT3R", NULL, "OUT3R" },
{ "AEC Loopback", "SPKOUTL", "OUT4L" },
{ "SPKOUTLN", NULL, "OUT4L" },
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3938fb1..53bbfac 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
- aif1 |= WM8904_AIF_LRCLK_INV;
+ aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
case SND_SOC_DAIFMT_DSP_A:
aif1 |= 0x3;
break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2..0f17ed3 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8962_CLOCKING_4,
WM8962_SYSCLK_RATE_MASK, clocking4);
+ /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK.
+ * So we here provisionally enable it and then disable it afterward
+ * if current bias_level hasn't reached SND_SOC_BIAS_ON.
+ */
+ if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA);
+
dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
+
+ if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_SYSCLK_ENA_MASK, 0);
+
if (dspclk < 0) {
dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
return;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9..4fbcab6 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
return ret;
/* Wait for the RAM to start, should be near instantaneous */
- count = 0;
- do {
+ for (count = 0; count < 10; ++count) {
ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
&val);
if (ret != 0)
return ret;
- } while (!(val & ADSP2_RAM_RDY) && ++count < 10);
+
+ if (val & ADSP2_RAM_RDY)
+ break;
+
+ msleep(1);
+ }
if (!(val & ADSP2_RAM_RDY)) {
adsp_err(dsp, "Failed to start DSP RAM\n");
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 61e4885..3fd76bc 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
break;
}
- dapm->bias_level = level;
-
return 0;
}
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index cbc9c96..41949af 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
}
}
+static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
+{
+ unsigned int i;
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
+ i++) {
+ if (!pcm->chan[i])
+ continue;
+ dma_release_channel(pcm->chan[i]);
+ if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+ break;
+ }
+}
+
/**
* snd_dmaengine_pcm_register - Register a dmaengine based PCM device
* @dev: The parent device for the PCM device
@@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev,
const struct snd_dmaengine_pcm_config *config, unsigned int flags)
{
struct dmaengine_pcm *pcm;
+ int ret;
pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
if (!pcm)
@@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev,
dmaengine_pcm_request_chan_of(pcm, dev);
if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
- return snd_soc_add_platform(dev, &pcm->platform,
+ ret = snd_soc_add_platform(dev, &pcm->platform,
&dmaengine_no_residue_pcm_platform);
else
- return snd_soc_add_platform(dev, &pcm->platform,
+ ret = snd_soc_add_platform(dev, &pcm->platform,
&dmaengine_pcm_platform);
+ if (ret)
+ goto err_free_dma;
+
+ return 0;
+
+err_free_dma:
+ dmaengine_pcm_release_chan(pcm);
+ kfree(pcm);
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register);
@@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
{
struct snd_soc_platform *platform;
struct dmaengine_pcm *pcm;
- unsigned int i;
platform = snd_soc_lookup_platform(dev);
if (!platform)
@@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
pcm = soc_platform_to_pcm(platform);
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
- if (pcm->chan[i]) {
- dma_release_channel(pcm->chan[i]);
- if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
- break;
- }
- }
-
snd_soc_remove_platform(platform);
+ dmaengine_pcm_release_chan(pcm);
kfree(pcm);
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 11a90cd..891b9a9 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -600,12 +600,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
/* apply codec digital mute */
- if (!codec->active)
+ if ((playback && codec_dai->playback_active == 1) ||
+ (!playback && codec_dai->capture_active == 1))
snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
/* free any machine hw params */
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 364bf6a9..8c819f8 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
unsigned int fmt)
{
struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- unsigned int mask, val;
+ unsigned int mask = 0, val = 0;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
@@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+ mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+ val |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 08bc693..8c7c102 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
{
struct device *dev = dai->dev;
struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
- unsigned int mask, val;
+ unsigned int mask = 0, val = 0;
int ret, spdifclock;
- mask = TEGRA20_SPDIF_CTRL_PACK |
- TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+ mask |= TEGRA20_SPDIF_CTRL_PACK |
+ TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- val = TEGRA20_SPDIF_CTRL_PACK |
- TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+ val |= TEGRA20_SPDIF_CTRL_PACK |
+ TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
break;
default:
return -EINVAL;
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 231a785..02247fe 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
unsigned int fmt)
{
struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- unsigned int mask, val;
+ unsigned int mask = 0, val = 0;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
@@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+ mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+ val |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 3454262..f4b12c2 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1603,7 +1603,7 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer)
return err;
}
- return err;
+ return 0;
}
int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
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