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-rw-r--r--sound/core/compress_offload.c8
-rw-r--r--sound/pci/hda/hda_codec.c12
-rw-r--r--sound/pci/hda/hda_codec.h1
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c6
-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/card.c4
-rw-r--r--sound/usb/endpoint.c24
-rw-r--r--sound/usb/endpoint.h3
-rw-r--r--sound/usb/pcm.c70
21 files changed, 115 insertions, 79 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index ec2118d0e27..eb60cb8dbb8 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 f560051a949..1c65cc5e3a3 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec)
kfree(codec);
}
+static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec,
+ hda_nid_t fg, unsigned int power_state);
+
static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state);
@@ -1317,6 +1320,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
AC_VERB_GET_SUBSYSTEM_ID, 0);
}
+ codec->epss = snd_hda_codec_get_supported_ps(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_EPSS);
+
/* power-up all before initialization */
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
@@ -2346,6 +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;
@@ -2361,7 +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);
@@ -3543,8 +3550,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
/* this delay seems necessary to avoid click noise at power-down */
if (power_state == AC_PWRST_D3) {
/* transition time less than 10ms for power down */
- bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS);
- msleep(epss ? 10 : 100);
+ msleep(codec->epss ? 10 : 100);
}
/* repeat power states setting at most 10 times*/
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 7fbc1bcaf1a..e5a7e19a807 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -862,6 +862,7 @@ struct hda_codec {
unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
unsigned int no_jack_detect:1; /* Machine has no jack-detection */
unsigned int pcm_format_first:1; /* PCM format must be set first */
+ unsigned int epss:1; /* supporting EPSS? */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
int power_transition; /* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 60882c62f18..c4763c52eaf 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 ea5775a1a7d..3d4722f0a1c 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
};
@@ -4543,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec,
struct auto_pin_cfg *cfg = &spec->autocfg;
int i;
+ if (cfg->speaker_outs == 0)
+ return;
+
for (i = 0; i < cfg->line_outs; i++) {
if (presence)
break;
@@ -5531,6 +5534,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e);
}
+ codec->epss = 0; /* longer delay needed for D3 */
codec->no_trigger_sense = 1;
codec->spec = spec;
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 764cc93dbca..075d5aa1fee 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 5c9cacaf2d5..1cf7a32d1b2 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 8f726c063f4..115a4030181 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 0013afe48e6..dc4262eea4b 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 fb21b17f17f..199408ec426 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 009533ab8d1..df65f98211e 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 f3ebc38c10f..b70964ea448 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 dd7c49fafd7..f90139b5f50 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 97c2cac8e92..8c7f2372944 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 e463529b38b..76cb1b363b7 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 5658bcec193..8d6900c1ee4 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 5c472f335a6..eb85113d472 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/card.c b/sound/usb/card.c
index d5b5c3388e2..4a469f0cb6d 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev,
struct snd_usb_audio *chip)
{
struct snd_card *card;
- struct list_head *p;
+ struct list_head *p, *n;
if (chip == (void *)-1L)
return;
@@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev,
snd_usb_stream_disconnect(p);
}
/* release the endpoint resources */
- list_for_each(p, &chip->ep_list) {
+ list_for_each_safe(p, n, &chip->ep_list) {
snd_usb_endpoint_free(p);
}
/* release the midi resources */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index c4118120268..d6e2bb49c59 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -141,7 +141,7 @@ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep)
*
* For implicit feedback, next_packet_size() is unused.
*/
-static int next_packet_size(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
{
unsigned long flags;
int ret;
@@ -177,15 +177,6 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep,
ep->retire_data_urb(ep->data_subs, urb);
}
-static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep,
- struct snd_urb_ctx *ctx)
-{
- int i;
-
- for (i = 0; i < ctx->packets; ++i)
- ctx->packet_size[i] = next_packet_size(ep);
-}
-
/*
* Prepare a PLAYBACK urb for submission to the bus.
*/
@@ -370,7 +361,6 @@ static void snd_complete_urb(struct urb *urb)
goto exit_clear;
}
- prepare_outbound_urb_sizes(ep, ctx);
prepare_outbound_urb(ep, ctx);
} else {
retire_inbound_urb(ep, ctx);
@@ -799,7 +789,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
/**
* snd_usb_endpoint_start: start an snd_usb_endpoint
*
- * @ep: the endpoint to start
+ * @ep: the endpoint to start
+ * @can_sleep: flag indicating whether the operation is executed in
+ * non-atomic context
*
* A call to this function will increment the use count of the endpoint.
* In case it is not already running, the URBs for this endpoint will be
@@ -809,7 +801,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
*
* Returns an error if the URB submission failed, 0 in all other cases.
*/
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep)
{
int err;
unsigned int i;
@@ -821,6 +813,11 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
if (++ep->use_count != 1)
return 0;
+ /* just to be sure */
+ deactivate_urbs(ep, 0, can_sleep);
+ if (can_sleep)
+ wait_clear_urbs(ep);
+
ep->active_mask = 0;
ep->unlink_mask = 0;
ep->phase = 0;
@@ -850,7 +847,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
goto __error;
if (usb_pipeout(ep->pipe)) {
- prepare_outbound_urb_sizes(ep, urb->context);
prepare_outbound_urb(ep, urb->context);
} else {
prepare_inbound_urb(ep, urb->context);
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index ee2723fb174..cbbbdf226d6 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -13,7 +13,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
struct audioformat *fmt,
struct snd_usb_endpoint *sync_ep);
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep);
void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
int force, int can_sleep, int wait);
int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
@@ -21,6 +21,7 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep);
void snd_usb_endpoint_free(struct list_head *head);
int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep);
void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
struct snd_usb_endpoint *sender,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 62ec808ed79..f782ce19bf5 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -212,7 +212,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
}
}
-static int start_endpoints(struct snd_usb_substream *subs)
+static int start_endpoints(struct snd_usb_substream *subs, int can_sleep)
{
int err;
@@ -225,7 +225,7 @@ static int start_endpoints(struct snd_usb_substream *subs)
snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep);
ep->data_subs = subs;
- err = snd_usb_endpoint_start(ep);
+ err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
return err;
@@ -236,10 +236,25 @@ static int start_endpoints(struct snd_usb_substream *subs)
!test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) {
struct snd_usb_endpoint *ep = subs->sync_endpoint;
+ if (subs->data_endpoint->iface != subs->sync_endpoint->iface ||
+ subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) {
+ err = usb_set_interface(subs->dev,
+ subs->sync_endpoint->iface,
+ subs->sync_endpoint->alt_idx);
+ if (err < 0) {
+ snd_printk(KERN_ERR
+ "%d:%d:%d: cannot set interface (%d)\n",
+ subs->dev->devnum,
+ subs->sync_endpoint->iface,
+ subs->sync_endpoint->alt_idx, err);
+ return -EIO;
+ }
+ }
+
snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep);
ep->sync_slave = subs->data_endpoint;
- err = snd_usb_endpoint_start(ep);
+ err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
return err;
@@ -544,13 +559,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
subs->last_frame_number = 0;
runtime->delay = 0;
- /* clear the pending deactivation on the target EPs */
- deactivate_endpoints(subs);
-
/* for playback, submit the URBs now; otherwise, the first hwptr_done
* updates for all URBs would happen at the same time when starting */
if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
- return start_endpoints(subs);
+ return start_endpoints(subs, 1);
return 0;
}
@@ -1032,6 +1044,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
struct urb *urb)
{
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ struct snd_usb_endpoint *ep = subs->data_endpoint;
struct snd_urb_ctx *ctx = urb->context;
unsigned int counts, frames, bytes;
int i, stride, period_elapsed = 0;
@@ -1043,7 +1056,11 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
urb->number_of_packets = 0;
spin_lock_irqsave(&subs->lock, flags);
for (i = 0; i < ctx->packets; i++) {
- counts = ctx->packet_size[i];
+ if (ctx->packet_size[i])
+ counts = ctx->packet_size[i];
+ else
+ counts = snd_usb_endpoint_next_packet_size(ep);
+
/* set up descriptor */
urb->iso_frame_desc[i].offset = frames * stride;
urb->iso_frame_desc[i].length = counts * stride;
@@ -1094,7 +1111,16 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
subs->hwptr_done += bytes;
if (subs->hwptr_done >= runtime->buffer_size * stride)
subs->hwptr_done -= runtime->buffer_size * stride;
+
+ /* update delay with exact number of samples queued */
+ runtime->delay = subs->last_delay;
runtime->delay += frames;
+ subs->last_delay = runtime->delay;
+
+ /* realign last_frame_number */
+ subs->last_frame_number = usb_get_current_frame_number(subs->dev);
+ subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
+
spin_unlock_irqrestore(&subs->lock, flags);
urb->transfer_buffer_length = bytes;
if (period_elapsed)
@@ -1112,12 +1138,32 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
int stride = runtime->frame_bits >> 3;
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);
- if (processed > runtime->delay)
- runtime->delay = 0;
+ est_delay = snd_usb_pcm_delay(subs, runtime->rate);
+ /* update delay with exact number of samples played */
+ if (processed > subs->last_delay)
+ subs->last_delay = 0;
else
- runtime->delay -= processed;
+ subs->last_delay -= processed;
+ runtime->delay = subs->last_delay;
+
+ /*
+ * Report when delay estimate is off by more than 2ms.
+ * The error should be lower than 2ms since the estimate relies
+ * on two reads of a counter updated every ms.
+ */
+ if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
+ snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
+ est_delay, subs->last_delay);
+
spin_unlock_irqrestore(&subs->lock, flags);
}
@@ -1175,7 +1221,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- err = start_endpoints(subs);
+ err = start_endpoints(subs, 0);
if (err < 0)
return err;
generated by cgit v1.2.3-70-g09d2 (git 2.51.2) at 2025-11-04 08:50:54 +0000