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:
  ASoC: tpa6130a2: Get rid of compile warning from tpa6130a2_power
  ALSA: hda - MacBookAir3,1(3,2) alsa support
  ASoC: fix the building issue of missing codec field in 'struct snd_soc_card'
  ALSA: usb-audio - Support for Power/Status LED on Creative USB X-Fi S51
  ALSA: asihpi - Unsafe memory management when allocating control cache
  ASoC: Update WARN uses in wm_hubs
  ASoC: Include cx20442 to SND_SOC_ALL_CODECS
  ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740
  ASoC: Remove volatility from WM8900 POWER1 register
  ALSA: lx6464es - make 1 bit signed bitfield unsigned
  ALSA: cs46xx memory management fixes for cs46xx_dsp_spos_create()
  ALSA: usb - driver neglects kmalloc return value check and may deref NULL
  ASoC: tpa6130a2: Fix unbalanced regulator disables
  ASoC: tlv320dac33: Mode1 FIFO auto configuration fix
  ASoC: tlv320dac33: Limit the US_TO_SAMPLES macro
  ASoC: tlv320dac33: Error handling for broken chip
  ASoC: Check return value of struct_strtoul() in pmdown_time_set()

sound/pci/asihpi/hpi6000.c

@@ -625,6 +625,8 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao,
 			control_cache_size, (struct hpi_control_cache_info *)
 			&phw->control_cache[0]
 			);
+		if (!phw->p_cache)
+			pao->has_control_cache = 0;
 	} else
 		pao->has_control_cache = 0;
 

sound/pci/asihpi/hpi6205.c

@@ -644,6 +644,8 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 				interface->control_cache.size_in_bytes,
 				(struct hpi_control_cache_info *)
 				p_control_cache_virtual);
+			if (!phw->p_cache)
+				err = HPI_ERROR_MEMORY_ALLOC;
 		}
 		if (!err) {
 			err = hpios_locked_mem_get_phys_addr(&phw->

sound/pci/asihpi/hpicmn.c

@@ -571,14 +571,20 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32
 {
 	struct hpi_control_cache *p_cache =
 		kmalloc(sizeof(*p_cache), GFP_KERNEL);
+	if (!p_cache)
+		return NULL;
+	p_cache->p_info =
+		kmalloc(sizeof(*p_cache->p_info) * number_of_controls,
+			GFP_KERNEL);
+	if (!p_cache->p_info) {
+		kfree(p_cache);
+		return NULL;
+	}
 	p_cache->cache_size_in_bytes = size_in_bytes;
 	p_cache->control_count = number_of_controls;
 	p_cache->p_cache =
 		(struct hpi_control_cache_single *)pDSP_control_buffer;
 	p_cache->init = 0;
-	p_cache->p_info =
-		kmalloc(sizeof(*p_cache->p_info) * p_cache->control_count,
-		GFP_KERNEL);
 	return p_cache;
 }
 

sound/pci/cs46xx/dsp_spos.c

@@ -229,35 +229,21 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip)
 		return NULL;
 
 	/* better to use vmalloc for this big table */
-	ins->symbol_table.nsymbols = 0;
 	ins->symbol_table.symbols = vmalloc(sizeof(struct dsp_symbol_entry) *
 					    DSP_MAX_SYMBOLS);
-	ins->symbol_table.highest_frag_index = 0;
-
-	if (ins->symbol_table.symbols == NULL) {
+	ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL);
+	ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL);
+	if (!ins->symbol_table.symbols || !ins->code.data || !ins->modules) {
 		cs46xx_dsp_spos_destroy(chip);
 		goto error;
 	}
-
+	ins->symbol_table.nsymbols = 0;
+	ins->symbol_table.highest_frag_index = 0;
 	ins->code.offset = 0;
 	ins->code.size = 0;
-	ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL);
-
-	if (ins->code.data == NULL) {
-		cs46xx_dsp_spos_destroy(chip);
-		goto error;
-	}
-
 	ins->nscb = 0;
 	ins->ntask = 0;
-
 	ins->nmodules = 0;
-	ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL);
-
-	if (ins->modules == NULL) {
-		cs46xx_dsp_spos_destroy(chip);
-		goto error;
-	}
 
 	/* default SPDIF input sample rate
 	   to 48000 khz */
@@ -281,6 +267,9 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip)
 	return ins;
 
 error:
+	kfree(ins->modules);
+	kfree(ins->code.data);
+	vfree(ins->symbol_table.symbols);
 	kfree(ins);
 	return NULL;
 }

sound/pci/hda/patch_cirrus.c

@@ -1166,6 +1166,7 @@ static const char *cs420x_models[CS420X_MODELS] = {
 
 static struct snd_pci_quirk cs420x_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53),
+	SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55),
 	SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
 	SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55),
 	SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),

sound/pci/lx6464es/lx6464es.c

@@ -425,7 +425,7 @@ static int lx_pcm_hw_free(struct snd_pcm_substream *substream)
 static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream)
 {
 	struct snd_pcm_substream *substream = lx_stream->stream;
-	const int is_capture = lx_stream->is_capture;
+	const unsigned int is_capture = lx_stream->is_capture;
 
 	int err;
 
@@ -473,7 +473,7 @@ static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream)
 
 static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream)
 {
-	const int is_capture = lx_stream->is_capture;
+	const unsigned int is_capture = lx_stream->is_capture;
 	int err;
 
 	snd_printd(LXP "stopping: stopping stream\n");

sound/pci/lx6464es/lx6464es.h

@@ -60,7 +60,7 @@ struct lx_stream {
 	snd_pcm_uframes_t          frame_pos;
 	enum lx_stream_status      status; /* free, open, running, draining
 					    * pause */
-	int                        is_capture:1;
+	unsigned int               is_capture:1;
 };
 
 

sound/pci/lx6464es/lx_core.c

@@ -1152,7 +1152,7 @@ static int lx_interrupt_request_new_buffer(struct lx6464es *chip,
 					   struct lx_stream *lx_stream)
 {
 	struct snd_pcm_substream *substream = lx_stream->stream;
-	int is_capture = lx_stream->is_capture;
+	const unsigned int is_capture = lx_stream->is_capture;
 	int err;
 	unsigned long flags;
 

sound/soc/codecs/Kconfig

@@ -25,8 +25,9 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS42L51 if I2C
 	select SND_SOC_CS4270 if I2C
+	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
-	select SND_SOC_JZ4740 if SOC_JZ4740
+	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008

sound/soc/codecs/tlv320dac33.c

@@ -58,7 +58,7 @@
 	(1000000000 / ((rate * 1000) / samples))
 
 #define US_TO_SAMPLES(rate, us) \
-	(rate / (1000000 / us))
+	(rate / (1000000 / (us < 1000000 ? us : 1000000)))
 
 #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
 	((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate)))
@@ -200,7 +200,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 		      u8 *value)
 {
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-	int val;
+	int val, ret = 0;
 
 	*value = reg & 0xff;
 
@@ -210,6 +210,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 		if (val < 0) {
 			dev_err(codec->dev, "Read failed (%d)\n", val);
 			value[0] = dac33_read_reg_cache(codec, reg);
+			ret = val;
 		} else {
 			value[0] = val;
 			dac33_write_reg_cache(codec, reg, val);
@@ -218,7 +219,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 		value[0] = dac33_read_reg_cache(codec, reg);
 	}
 
-	return 0;
+	return ret;
 }
 
 static int dac33_write(struct snd_soc_codec *codec, unsigned int reg,
@@ -329,13 +330,18 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
 		    dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
 }
 
-static inline void dac33_read_id(struct snd_soc_codec *codec)
+static inline int dac33_read_id(struct snd_soc_codec *codec)
 {
+	int i, ret = 0;
 	u8 reg;
 
-	dac33_read(codec, DAC33_DEVICE_ID_MSB, &reg);
-	dac33_read(codec, DAC33_DEVICE_ID_LSB, &reg);
-	dac33_read(codec, DAC33_DEVICE_REV_ID, &reg);
+	for (i = 0; i < 3; i++) {
+		ret = dac33_read(codec, DAC33_DEVICE_ID_MSB + i, &reg);
+		if (ret < 0)
+			break;
+	}
+
+	return ret;
 }
 
 static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
@@ -1076,6 +1082,9 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 		/* Number of samples under i2c latency */
 		dac33->alarm_threshold = US_TO_SAMPLES(rate,
 						dac33->mode1_latency);
+		nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
+				dac33->alarm_threshold;
+
 		if (dac33->auto_fifo_config) {
 			if (period_size <= dac33->alarm_threshold)
 				/*
@@ -1086,6 +1095,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 				       ((dac33->alarm_threshold / period_size) +
 				       (dac33->alarm_threshold % period_size ?
 				       1 : 0));
+			else if (period_size > nsample_limit)
+				dac33->nsample = nsample_limit;
 			else
 				dac33->nsample = period_size;
 		} else {
@@ -1097,8 +1108,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 			 */
 			dac33->nsample_max = substream->runtime->buffer_size -
 						period_size;
-			nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
-					dac33->alarm_threshold;
+
 			if (dac33->nsample_max > nsample_limit)
 				dac33->nsample_max = nsample_limit;
 
@@ -1414,9 +1424,15 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
 		dev_err(codec->dev, "Failed to power up codec: %d\n", ret);
 		goto err_power;
 	}
-	dac33_read_id(codec);
+	ret = dac33_read_id(codec);
 	dac33_hard_power(codec, 0);
 
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to read chip ID: %d\n", ret);
+		ret = -ENODEV;
+		goto err_power;
+	}
+
 	/* Check if the IRQ number is valid and request it */
 	if (dac33->irq >= 0) {
 		ret = request_irq(dac33->irq, dac33_interrupt_handler,

sound/soc/codecs/tpa6130a2.c

@@ -119,13 +119,13 @@ static int tpa6130a2_power(int power)
 {
 	struct	tpa6130a2_data *data;
 	u8	val;
-	int	ret;
+	int	ret = 0;
 
 	BUG_ON(tpa6130a2_client == NULL);
 	data = i2c_get_clientdata(tpa6130a2_client);
 
 	mutex_lock(&data->mutex);
-	if (power) {
+	if (power && !data->power_state) {
 		/* Power on */
 		if (data->power_gpio >= 0)
 			gpio_set_value(data->power_gpio, 1);
@@ -153,7 +153,7 @@ static int tpa6130a2_power(int power)
 		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
 		val &= ~TPA6130A2_SWS;
 		tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
-	} else {
+	} else if (!power && data->power_state) {
 		/* set SWS */
 		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
 		val |= TPA6130A2_SWS;

sound/soc/codecs/wm8900.c

@@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg)
 {
 	switch (reg) {
 	case WM8900_REG_ID:
-	case WM8900_REG_POWER1:
 		return 1;
 	default:
 		return 0;
@@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec)
 		return -ENODEV;
 	}
 
-	/* Read back from the chip */
-	reg = snd_soc_read(codec, WM8900_REG_POWER1);
-	reg = (reg >> 12) & 0xf;
-	dev_info(codec->dev, "WM8900 revision %d\n", reg);
-
 	wm8900_reset(codec);
 
 	/* Turn the chip on */

sound/soc/codecs/wm_hubs.c

@@ -123,7 +123,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 			reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
 			break;
 		default:
-			WARN(1, "Unknown DCS readback method");
+			WARN(1, "Unknown DCS readback method\n");
 			break;
 		}
 

sound/soc/pxa/tosa.c

@@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);

sound/soc/soc-core.c

@@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev,
 {
 	struct snd_soc_pcm_runtime *rtd =
 			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	int ret;
 
-	strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	if (ret)
+		return ret;
 
 	return count;
 }

sound/usb/mixer_quirks.c

@@ -60,7 +60,7 @@ static const struct rc_config {
 	{ USB_ID(0x041e, 0x3000), 0, 1, 2, 1,  18, 0x0013 }, /* Extigy       */
 	{ USB_ID(0x041e, 0x3020), 2, 1, 6, 6,  18, 0x0013 }, /* Audigy 2 NX  */
 	{ USB_ID(0x041e, 0x3040), 2, 2, 6, 6,  2,  0x6e91 }, /* Live! 24-bit */
-	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi */
+	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 */
 	{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6,  2,  0x6e91 }, /* Toshiba SB0500 */
 };
 
@@ -183,6 +183,12 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	if (value > 1)
 		return -EINVAL;
 	changed = value != mixer->audigy2nx_leds[index];
+	if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042))
+		err = snd_usb_ctl_msg(mixer->chip->dev,
+			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
+			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+			      !value, 0, NULL, 0, 100);
+	else
 		err = snd_usb_ctl_msg(mixer->chip->dev,
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
 			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
@@ -225,8 +231,12 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer)
 	int i, err;
 
 	for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) {
+		/* USB X-Fi S51 doesn't have a CMSS LED */
+		if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0)
+			continue;
 		if (i > 1 && /* Live24ext has 2 LEDs only */
 			(mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+			 mixer->chip->usb_id == USB_ID(0x041e, 0x3042) ||
 			 mixer->chip->usb_id == USB_ID(0x041e, 0x3048)))
 			break; 
 		err = snd_ctl_add(mixer->chip->card,
@@ -365,6 +375,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 
 	if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) ||
 	    mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+	    mixer->chip->usb_id == USB_ID(0x041e, 0x3042) ||
 	    mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) {
 		if ((err = snd_audigy2nx_controls_create(mixer)) < 0)
 			return err;

sound/usb/pcm.c

@@ -676,8 +676,10 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime,
 	if (!needs_knot)
 		return 0;
 
-	subs->rate_list.count = count;
 	subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL);
+	if (!subs->rate_list.list)
+		return -ENOMEM;
+	subs->rate_list.count = count;
 	subs->rate_list.mask = 0;
 	count = 0;
 	list_for_each_entry(fp, &subs->fmt_list, list) {