Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop
  ALSA: Use %pV for snd_printk()
  ALSA: hda - Fix initialization of hp pins with master_mute in Realtek
  ALSA: hda - Fix invalid unsol tag for some alc262 model quirks
  ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register
  ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context
  ASoC: fsl: fix initialization of DMA buffers
  ASoC: WM8804 does not support sample rates below 32kHz
  ASoC: Fix WM8962 headphone volume update for use of advanced caches
  ASoC: Blackfin: bf5xx-ad1836: Fix codec device name
  ALSA: hda: Fix quirk for Dell Inspiron 910
  ASoC: AD1836: Fix setting the PCM format
  ASoC: Check for NULL register bank in snd_soc_get_cache_val()
  ASoC: Add missing break in WM8915 FLL source selection
  ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK
  ASoC: atmel_ssc: Don't try to free ssc if request failed

14 files changed, 74 insertions, 60 deletions

diff --git a/sound/core/misc.c b/sound/core/misc.c
index 2c41825c836e..eb9fe2e1d291 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -58,26 +58,6 @@ static const char *sanity_file_name(const char *path)
 	else
 		return path;
 }
-
-/* print file and line with a certain printk prefix */
-static int print_snd_pfx(unsigned int level, const char *path, int line,
-			 const char *format)
-{
-	const char *file = sanity_file_name(path);
-	char tmp[] = "<0>";
-	const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT;
-	int ret = 0;
-
-	if (format[0] == '<' && format[2] == '>') {
-		tmp[1] = format[1];
-		pfx = tmp;
-		ret = 1;
-	}
-	printk("%sALSA %s:%d: ", pfx, file, line);
-	return ret;
-}
-#else
-#define print_snd_pfx(level, path, line, format)	0
 #endif
 
 #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
@@ -85,15 +65,29 @@ void __snd_printk(unsigned int level, const char *path, int line,
 		  const char *format, ...)
 {
 	va_list args;
-	
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+	struct va_format vaf;
+	char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV";
+#endif
+
 #ifdef CONFIG_SND_DEBUG	
 	if (debug < level)
 		return;
 #endif
+
 	va_start(args, format);
-	if (print_snd_pfx(level, path, line, format))
-		format += 3; /* skip the printk level-prefix */
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+	vaf.fmt = format;
+	vaf.va = &args;
+	if (format[0] == '<' && format[2] == '>') {
+		memcpy(verbose_fmt, format, 3);
+		vaf.fmt = format + 3;
+	} else if (level)
+		memcpy(verbose_fmt, KERN_DEBUG, 3);
+	printk(verbose_fmt, sanity_file_name(path), line, &vaf);
+#else
 	vprintk(format, args);
+#endif
 	va_end(args);
 }
 EXPORT_SYMBOL_GPL(__snd_printk);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3e6b9a8539c2..694b9daf691f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3102,6 +3102,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
+	SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7a4e10002f56..43fcfbd32847 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1141,6 +1141,13 @@ static void update_speakers(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int on;
 
+	/* Control HP pins/amps depending on master_mute state;
+	 * in general, HP pins/amps control should be enabled in all cases,
+	 * but currently set only for master_mute, just to be safe
+	 */
+	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
+		    spec->autocfg.hp_pins, spec->master_mute, true);
+
 	if (!spec->automute)
 		on = 0;
 	else
@@ -6201,11 +6208,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = {
 /* update HP, line and mono out pins according to the master switch */
 static void alc260_hp_master_update(struct hda_codec *codec)
 {
-	struct alc_spec *spec = codec->spec;
-
-	/* change HP pins */
-	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
-		    spec->autocfg.hp_pins, spec->master_mute, true);
 	update_speakers(codec);
 }
 
@@ -11924,7 +11926,7 @@ static const struct hda_verb alc262_nec_verbs[] = {
  *  0x1b = port replicator headphone out
  */
 
-#define ALC_HP_EVENT	0x37
+#define ALC_HP_EVENT	ALC880_HP_EVENT
 
 static const struct hda_verb alc262_fujitsu_unsol_verbs[] = {
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
@@ -13860,6 +13862,7 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
 						ALC268_ACER_ASPIRE_ONE),
 	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
+	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
 	SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
 			"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
 	/* almost compatible with toshiba but with optional digital outs;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 7fbfa051f6e1..eda955b15834 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id)
 	if (IS_ERR(ssc))
 		pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
 			PTR_ERR(ssc));
-	else
+	else {
 		ssc_pdev->dev.parent = &(ssc->pdev->dev);
-	ssc_free(ssc);
+		ssc_free(ssc);
+	}
 
 	ret = platform_device_add(ssc_pdev);
 	if (ret < 0)
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index ea4951cf5526..f79d1655e035 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
 		.cpu_dai_name = "bfin-tdm.0",
 		.codec_dai_name = "ad1836-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad1836.0",
+		.codec_name = "spi0.4",
 		.ops = &bf5xx_ad1836_ops,
 	},
 	{
@@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
 		.cpu_dai_name = "bfin-tdm.1",
 		.codec_dai_name = "ad1836-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad1836.0",
+		.codec_name = "spi0.4",
 		.ops = &bf5xx_ad1836_ops,
 	},
 };
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index ab63d52e36e1..754c496412bd 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 	/* bit size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		word_len = 3;
+		word_len = AD1836_WORD_LEN_16;
 		break;
 	case SNDRV_PCM_FORMAT_S20_3LE:
-		word_len = 1;
+		word_len = AD1836_WORD_LEN_20;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 	case SNDRV_PCM_FORMAT_S32_LE:
-		word_len = 0;
+		word_len = AD1836_WORD_LEN_24;
 		break;
 	}
 
-	snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
-		AD1836_DAC_WORD_LEN_MASK, word_len);
+	snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK,
+		word_len << AD1836_DAC_WORD_LEN_OFFSET);
 
-	snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
-		AD1836_ADC_WORD_LEN_MASK, word_len);
+	snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK,
+		word_len << AD1836_ADC_WORD_OFFSET);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 845596717fdf..9d6a3f8f8aaf 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -25,6 +25,7 @@
 #define AD1836_DAC_SERFMT_PCK256       (0x4 << 5)
 #define AD1836_DAC_SERFMT_PCK128       (0x5 << 5)
 #define AD1836_DAC_WORD_LEN_MASK       0x18
+#define AD1836_DAC_WORD_LEN_OFFSET     3
 
 #define AD1836_DAC_CTRL2               1
 #define AD1836_DACL1_MUTE              0
@@ -51,6 +52,7 @@
 #define AD1836_ADCL2_MUTE 		2
 #define AD1836_ADCR2_MUTE 		3
 #define AD1836_ADC_WORD_LEN_MASK       0x30
+#define AD1836_ADC_WORD_OFFSET         5
 #define AD1836_ADC_SERFMT_MASK	       (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
@@ -60,4 +62,8 @@
 
 #define AD1836_NUM_REGS                16
 
+#define AD1836_WORD_LEN_24 0x0
+#define AD1836_WORD_LEN_20 0x1
+#define AD1836_WORD_LEN_16 0x2
+
 #endif
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6785688f8806..9a5e67c5a6bd 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = {
 #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+		      SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+		      SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+		      SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
 static struct snd_soc_dai_driver wm8804_dai = {
 	.name = "wm8804-spdif",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000,
+		.rates = WM8804_RATES,
 		.formats = WM8804_FORMATS,
 	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000,
+		.rates = WM8804_RATES,
 		.formats = WM8804_FORMATS,
 	},
 	.ops = &wm8804_dai_ops,
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index a0b1a7278284..e2ab4fac2819 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
 	int old;
 
 	/* Disable SYSCLK while we reconfigure */
-	old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1);
+	old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
 			    WM8915_SYSCLK_ENA, 0);
 
@@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 		break;
 	case WM8915_FLL_MCLK2:
 		reg = 1;
+		break;
 	case WM8915_FLL_DACLRCLK1:
 		reg = 2;
 		break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index f90ae427242b..5e05eed96c38 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
 		return 0;
 
 	/* If the left PGA is enabled hit that VU bit... */
-	if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA)
+	if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
 		return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
 				     reg_cache[WM8962_HPOUTL_VOLUME]);
 
 	/* ...otherwise the right.  The VU is stereo. */
-	if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA)
+	if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
 		return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
 				     reg_cache[WM8962_HPOUTR_VOLUME]);
 
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 15dac0f20cd8..6680c0b4d203 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
 	 * should allocate a DMA buffer only for the streams that are valid.
 	 */
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[0].substream) {
 		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
 			fsl_dma_hardware.buffer_bytes_max,
 			&pcm->streams[0].substream->dma_buffer);
@@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
 		}
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[1].substream) {
 		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
 			fsl_dma_hardware.buffer_bytes_max,
 			&pcm->streams[1].substream->dma_buffer);
 		if (ret) {
-			snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
 			dev_err(card->dev, "can't alloc capture dma buffer\n");
+			snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
 			return ret;
 		}
 	}
@@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 	dma_private->ld_buf_phys = ld_buf_phys;
 	dma_private->dma_buf_phys = substream->dma_buffer.addr;
 
-	ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private);
+	ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio",
+			  dma_private);
 	if (ret) {
 		dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n",
 			dma_private->irq, ret);
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index ffa09b3b2caa..992a732b5211 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s)
 	if (!i2s)
 		return false;
 
-	active = readl(i2s->addr + I2SMOD);
+	active = readl(i2s->addr + I2SCON);
 
 	if (is_secondary(i2s))
 		active &= CON_TXSDMA_ACTIVE;
@@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s)
 	if (!i2s)
 		return false;
 
-	active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE;
+	active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE;
 
 	return active ? true : false;
 }
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 06b7b81a1601..c005ceb70c9d 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -466,6 +466,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx,
 static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx,
 		unsigned int word_size)
 {
+	if (!base)
+		return -1;
+
 	switch (word_size) {
 	case 1: {
 		const u8 *cache = base;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 776e6f418306..32ab7fc4579a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
-	struct snd_soc_dapm_widget *w)
+static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 {
+	struct snd_soc_dapm_context *dapm = w->dapm;
 	int i, ret = 0;
 	size_t name_len, prefix_len;
 	struct snd_soc_dapm_path *path;
@@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm mux control */
-static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
-	struct snd_soc_dapm_widget *w)
+static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 {
+	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_dapm_path *path = NULL;
 	struct snd_kcontrol *kcontrol;
 	struct snd_card *card = dapm->card->snd_card;
@@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm volume control */
-static int dapm_new_pga(struct snd_soc_dapm_context *dapm,
-	struct snd_soc_dapm_widget *w)
+static int dapm_new_pga(struct snd_soc_dapm_widget *w)
 {
 	if (w->num_kcontrols)
 		dev_err(w->dapm->dev,
@@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 		case snd_soc_dapm_mixer:
 		case snd_soc_dapm_mixer_named_ctl:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_mixer(dapm, w);
+			dapm_new_mixer(w);
 			break;
 		case snd_soc_dapm_mux:
 		case snd_soc_dapm_virt_mux:
 		case snd_soc_dapm_value_mux:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_mux(dapm, w);
+			dapm_new_mux(w);
 			break;
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_aif_out:
@@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_out_drv:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_pga(dapm, w);
+			dapm_new_pga(w);
 			break;
 		case snd_soc_dapm_input:
 		case snd_soc_dapm_output: