Merge tag 'v3.6-rc6' into for-3.7

Linux 3.6-rc6 has all our bug fixes.

Conflicts (trivial overlap):
	sound/soc/omap/am3517evm.c

20 files changed, 114 insertions, 78 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 f560051a949e..1c65cc5e3a31 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 7fbc1bcaf1a9..e5a7e19a8071 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 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 ea5775a1a7db..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
 };
 
@@ -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 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 b79578e7e10c..c167c896eaee 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -492,7 +492,7 @@ 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 d89e343ff108..bc955999c8aa 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -661,7 +661,7 @@ 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,
@@ -672,7 +672,7 @@ 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,
@@ -694,14 +694,14 @@ 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 2b9766d55384..7c8df52a8d9d 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/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 a18d115bc507..873e6e76ee87 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -313,8 +313,11 @@ 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 36be11e47ad6..1b7c2f58ce13 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;
@@ -687,15 +686,14 @@ 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;
@@ -707,7 +705,7 @@ 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;
@@ -718,14 +716,6 @@ 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,
@@ -734,11 +724,6 @@ 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 d5b5c3388e28..4a469f0cb6d4 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 c41181202688..d6e2bb49c59c 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 ee2723fb174f..cbbbdf226d66 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 62ec808ed792..f782ce19bf5a 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;