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-rw-r--r--sound/arm/aaci.c6
-rw-r--r--sound/pci/hda/patch_realtek.c1
-rw-r--r--sound/soc/codecs/tlv320aic23.c5
-rw-r--r--sound/soc/omap/omap3evm.c2
-rw-r--r--sound/soc/omap/omap3pandora.c3
-rw-r--r--sound/soc/soc-dapm.c20
-rw-r--r--sound/usb/usbmixer.c9
7 files changed, 37 insertions, 9 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1f0f8213e2d5..6c160a038b23 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -504,6 +504,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
int err;
aaci_pcm_hw_free(substream);
+ if (aacirun->pcm_open) {
+ snd_ac97_pcm_close(aacirun->pcm);
+ aacirun->pcm_open = 0;
+ }
err = devdma_hw_alloc(NULL, substream,
params_buffer_bytes(params));
@@ -517,7 +521,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
else
err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
params_channels(params),
- aacirun->pcm->r[1].slots);
+ aacirun->pcm->r[0].slots);
if (err)
goto out;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 84a52efdb2d6..70583719282b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11462,6 +11462,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 0b8dcb5cd729..90a0264f7538 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = {
#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
static const unsigned short sr_valid_mask[] = {
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
- LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
LOWER_GROUP, /* Usb, bosr - 0*/
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
UPPER_GROUP, /* Usb, bosr - 1*/
};
/*
@@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
- int i;
u16 reg;
/* Sync reg_cache with the hardware */
- for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+ for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 9114c263077b..13aa380de162 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit);
MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index ad219aaf7cb8..0cd06f5dd356 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
* |P| <--- TWL4030 <--------- Line In and MICs
*/
static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
- SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
0, 0, NULL, 0, omap3pandora_hp_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
snd_soc_dapm_nc_pin(codec, "CARKITR");
snd_soc_dapm_nc_pin(codec, "HFL");
snd_soc_dapm_nc_pin(codec, "HFR");
+ snd_soc_dapm_nc_pin(codec, "VIBRA");
ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
ARRAY_SIZE(omap3pandora_out_dapm_widgets));
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d89f6dc00908..66d4c165f99b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
if (!w->power_check)
continue;
- power = w->power_check(w);
- if (power)
- sys_power = 1;
+ /* If we're suspending then pull down all the
+ * power. */
+ switch (event) {
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ power = 0;
+ break;
+
+ default:
+ power = w->power_check(w);
+ if (power)
+ sys_power = 1;
+ break;
+ }
if (w->power == power)
continue;
@@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
case SND_SOC_DAPM_STREAM_RESUME:
sys_power = 1;
break;
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ sys_power = 0;
+ break;
case SND_SOC_DAPM_STREAM_NOP:
sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY;
+ break;
default:
break;
}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 9efcfd08d747..c998220b99c6 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig
channels = (ftr[0] - 7) / csize - 1;
master_bits = snd_usb_combine_bytes(ftr + 6, csize);
+ /* master configuration quirks */
+ switch (state->chip->usb_id) {
+ case USB_ID(0x08bb, 0x2702):
+ snd_printk(KERN_INFO
+ "usbmixer: master volume quirk for PCM2702 chip\n");
+ /* disable non-functional volume control */
+ master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1));
+ break;
+ }
if (channels > 0)
first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize);
else
generated by cgit v1.2.3 (git 2.46.0) at 2025-09-06 02:58:18 +0000