Merge branch 'fix/asoc' into for-linus

11 files changed, 47 insertions, 31 deletions

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: