summaryrefslogtreecommitdiffstats
path: root/sound
diff options
context:
space:
mode:
Diffstat (limited to 'sound')
-rw-r--r--sound/core/compress_offload.c8
-rw-r--r--sound/pci/hda/hda_codec.c2
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c2
-rw-r--r--sound/pci/ice1712/prodigy_hifi.c3
-rw-r--r--sound/soc/codecs/arizona.c2
-rw-r--r--sound/soc/codecs/mc13783.c8
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c2
-rw-r--r--sound/soc/omap/am3517evm.c2
-rw-r--r--sound/soc/samsung/dma.c8
-rw-r--r--sound/soc/soc-dapm.c5
-rw-r--r--sound/soc/spear/spear_pcm.c2
-rw-r--r--sound/soc/tegra/tegra_alc5632.c1
-rw-r--r--sound/soc/tegra/tegra_pcm.c4
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.c25
-rw-r--r--sound/usb/pcm.c6
17 files changed, 36 insertions, 48 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index ec2118d0e27a..eb60cb8dbb8a 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f)
int maj = imajor(inode);
int ret;
- if (f->f_flags & O_WRONLY)
+ if ((f->f_flags & O_ACCMODE) == O_WRONLY)
dirn = SND_COMPRESS_PLAYBACK;
- else if (f->f_flags & O_RDONLY)
+ else if ((f->f_flags & O_ACCMODE) == O_RDONLY)
dirn = SND_COMPRESS_CAPTURE;
- else {
- pr_err("invalid direction\n");
+ else
return -EINVAL;
- }
if (maj == snd_major)
compr = snd_lookup_minor_data(iminor(inode),
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index f25c24c743f9..1c65cc5e3a31 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2353,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
}
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
+ memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
snd_hda_jack_tbl_clear(codec);
codec->proc_widget_hook = NULL;
codec->spec = NULL;
@@ -2368,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec)
codec->num_pcms = 0;
codec->pcm_info = NULL;
codec->preset = NULL;
- memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
codec->slave_dig_outs = NULL;
codec->spdif_status_reset = 0;
module_put(codec->owner);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 60882c62f180..c4763c52eaf6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2701,6 +2701,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF),
+ SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6f806d3e56bb..3d4722f0a1ca 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1075,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = {
static const char * const slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
- "Headphone", "Speaker", "IEC958",
+ "Headphone", "Speaker", "IEC958", "PCM",
NULL
};
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 764cc93dbca4..075d5aa1fee0 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
}
static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
{
@@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
.info = ak4396_dac_vol_info,
.get = ak4396_dac_vol_get,
.put = ak4396_dac_vol_put,
- .tlv = { .p = db_scale_wm_dac },
+ .tlv = { .p = ak4396_db_scale },
},
};
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 5c9cacaf2d52..1cf7a32d1b21 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = {
940800,
1411200,
1881600,
- 2882400,
+ 2822400,
3763200,
5644800,
7526400,
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 8f726c063f42..115a40301810 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -659,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = {
.id = MC13783_ID_STEREO_DAC,
.playback = {
.stream_name = "Playback",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = MC13783_FORMATS,
@@ -670,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = {
.id = MC13783_ID_STEREO_CODEC,
.capture = {
.stream_name = "Capture",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = MC13783_RATES_RECORD,
.formats = MC13783_FORMATS,
@@ -692,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = {
.id = MC13783_ID_SYNC,
.playback = {
.stream_name = "Playback",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = MC13783_FORMATS,
},
.capture = {
.stream_name = "Capture",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = MC13783_RATES_RECORD,
.formats = MC13783_FORMATS,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 0013afe48e66..dc4262eea4b7 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = {
{ 14, 0x0000 }, /* R14 - Power Management 2 */
{ 15, 0x0000 }, /* R15 - Power Management 3 */
{ 18, 0x0000 }, /* R18 - Power Management 6 */
- { 19, 0x945E }, /* R20 - Clock Rates 0 */
+ { 20, 0x945E }, /* R20 - Clock Rates 0 */
{ 21, 0x0C05 }, /* R21 - Clock Rates 1 */
{ 22, 0x0006 }, /* R22 - Clock Rates 2 */
{ 24, 0x0050 }, /* R24 - Audio Interface 0 */
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index fb21b17f17f5..199408ec4261 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -94,7 +94,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "audmux internal port setup failed\n");
return ret;
}
- imx_audmux_v2_configure_port(ext_port,
+ ret = imx_audmux_v2_configure_port(ext_port,
IMX_AUDMUX_V2_PTCR_SYN,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
if (ret) {
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 009533ab8d18..df65f98211ec 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -59,7 +59,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream,
return ret;
}
- snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
+ ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n");
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index f3ebc38c10fe..b70964ea448c 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME,
+ SNDRV_PCM_INFO_MMAP_VALID,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_LE |
SNDRV_PCM_FMTBIT_U8 |
@@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->state |= ST_RUNNING;
prtd->params->ops->trigger(prtd->params->ch);
break;
case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
prtd->state &= ~ST_RUNNING;
prtd->params->ops->stop(prtd->params->ch);
break;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dd7c49fafd75..f90139b5f50d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -291,8 +291,11 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
if (dapm->codec->driver->set_bias_level)
ret = dapm->codec->driver->set_bias_level(dapm->codec,
level);
- } else
+ else
+ dapm->bias_level = level;
+ } else if (!card || dapm != &card->dapm) {
dapm->bias_level = level;
+ }
if (ret != 0)
goto out;
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 97c2cac8e92c..8c7f23729446 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -138,7 +138,7 @@ static void spear_pcm_free(struct snd_pcm *pcm)
continue;
buf = &substream->dma_buffer;
- if (!buf && !buf->area)
+ if (!buf || !buf->area)
continue;
dma_free_writecombine(pcm->card->dev, buf->bytes,
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index e463529b38bb..76cb1b363b71 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = {
.name = "Headset detection",
.report = SND_JACK_HEADSET,
.debounce_time = 150,
- .invert = 1,
};
static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = {
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 5658bcec1931..8d6900c1ee47 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
slave_config.dst_addr = dmap->addr;
- slave_config.src_maxburst = 0;
+ slave_config.dst_maxburst = 4;
} else {
slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
slave_config.src_addr = dmap->addr;
- slave_config.dst_maxburst = 0;
+ slave_config.src_maxburst = 4;
}
slave_config.slave_id = dmap->req_sel;
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 5c472f335a64..eb85113d472a 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -663,7 +663,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
struct ux500_msp **msp_p,
struct msp_i2s_platform_data *platform_data)
{
- int ret = 0;
struct resource *res = NULL;
struct i2s_controller *i2s_cont;
struct ux500_msp *msp;
@@ -685,15 +684,14 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
if (res == NULL) {
dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n",
__func__);
- ret = -ENOMEM;
- goto err_res;
+ return -ENOMEM;
}
- msp->registers = ioremap(res->start, (res->end - res->start + 1));
+ msp->registers = devm_ioremap(&pdev->dev, res->start,
+ resource_size(res));
if (msp->registers == NULL) {
dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__);
- ret = -ENOMEM;
- goto err_res;
+ return -ENOMEM;
}
msp->msp_state = MSP_STATE_IDLE;
@@ -705,7 +703,7 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
dev_err(&pdev->dev,
"%s: ERROR: Failed to allocate I2S-controller!\n",
__func__);
- goto err_i2s_cont;
+ return -ENOMEM;
}
i2s_cont->dev.parent = &pdev->dev;
i2s_cont->data = (void *)msp;
@@ -716,14 +714,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
msp->i2s_cont = i2s_cont;
return 0;
-
-err_i2s_cont:
- iounmap(msp->registers);
-
-err_res:
- devm_kfree(&pdev->dev, msp);
-
- return ret;
}
void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
@@ -732,11 +722,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id);
device_unregister(&msp->i2s_cont->dev);
- devm_kfree(&pdev->dev, msp->i2s_cont);
-
- iounmap(msp->registers);
-
- devm_kfree(&pdev->dev, msp);
}
MODULE_LICENSE("GPL v2");
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index fd5e982fc98c..f782ce19bf5a 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1140,6 +1140,12 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
int processed = urb->transfer_buffer_length / stride;
int est_delay;
+ /* ignore the delay accounting when procssed=0 is given, i.e.
+ * silent payloads are procssed before handling the actual data
+ */
+ if (!processed)
+ return;
+
spin_lock_irqsave(&subs->lock, flags);
est_delay = snd_usb_pcm_delay(subs, runtime->rate);
/* update delay with exact number of samples played */
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-26 01:42:48 +0100