Merge tag 'v3.6' into staging/for_v3.7

Linux 3.6

* tag 'v3.6': (562 commits)
  Linux 3.6
  vfs: dcache: fix deadlock in tree traversal
  mtdchar: fix offset overflow detection
  thp: avoid VM_BUG_ON page_count(page) false positives in __collapse_huge_page_copy
  iommu/amd: Fix wrong assumption in iommu-group specific code
  netdev: octeon: fix return value check in octeon_mgmt_init_phy()
  ALSA: snd-usb: fix next_packet_size calls for pause case
  inetpeer: fix token initialization
  qlcnic: Fix scheduling while atomic bug
  bnx2: Clean up remaining iounmap
  trivial select_parent documentation fix
  net: phy: smsc: Implement PHY config_init for LAN87xx
  smsc75xx: fix resume after device reset
  um: Preinclude include/linux/kern_levels.h
  um: Fix IPC on um
  netdev: pasemi: fix return value check in pasemi_mac_phy_init()
  team: fix return value check
  l2tp: fix return value check
  USB: Fix race condition when removing host controllers
  USB: ohci-at91: fix null pointer in ohci_hcd_at91_overcurrent_irq
  ...

19 files changed, 44 insertions, 50 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/wm2000.c b/sound/soc/codecs/wm2000.c
index 3fd5b29dc933..a3acb7a85f6a 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -702,7 +702,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg)
 }
 
 static const struct regmap_config wm2000_regmap = {
-	.reg_bits = 8,
+	.reg_bits = 16,
 	.val_bits = 8,
 
 	.max_register = WM2000_REG_IF_CTL,
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/endpoint.c b/sound/usb/endpoint.c
index d6e2bb49c59c..060dccb9ec75 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -197,7 +197,13 @@ static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
 			/* no data provider, so send silence */
 			unsigned int offs = 0;
 			for (i = 0; i < ctx->packets; ++i) {
-				int counts = ctx->packet_size[i];
+				int counts;
+
+				if (ctx->packet_size[i])
+					counts = ctx->packet_size[i];
+				else
+					counts = snd_usb_endpoint_next_packet_size(ep);
+
 				urb->iso_frame_desc[i].offset = offs * ep->stride;
 				urb->iso_frame_desc[i].length = counts * ep->stride;
 				offs += counts;
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 */