19 files changed, 103 insertions, 144 deletions

diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index f200d1cfc4bd..667f4215dfc0 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -26,7 +26,6 @@
 #include <sound/soc.h>
 
 #include "mpc5200_dma.h"
-#include "mpc5200_psc_ac97.h"
 
 #define DRV_NAME "efika-audio-fabric"
 
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 883087f2b092..84ef6385736c 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -64,7 +64,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops eukrea_tlv320_snd_ops = {
+static const struct snd_soc_ops eukrea_tlv320_snd_ops = {
 	.hw_params	= eukrea_tlv320_hw_params,
 };
 
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 1d82f68305c3..8cfffa70c144 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -368,7 +368,7 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
 	fsl_asrc_set_watermarks(pair, ASRC_INPUTFIFO_THRESHOLD,
 				ASRC_INPUTFIFO_THRESHOLD);
 
-	/* Configure the followings only for Ideal Ratio mode */
+	/* Configure the following only for Ideal Ratio mode */
 	if (!ideal)
 		return 0;
 
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index dc30d780f874..282d841840b1 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -76,7 +76,7 @@ static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream)
 			pair->dma_chan[!dir], runtime->dma_addr,
 			snd_pcm_lib_buffer_bytes(substream),
 			snd_pcm_lib_period_bytes(substream),
-			dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags);
+			dir == OUT ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, flags);
 	if (!pair->desc[!dir]) {
 		dev_err(dev, "failed to prepare slave DMA for Front-End\n");
 		return -ENOMEM;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 38bfd46f4ad8..809a069d490b 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -19,7 +19,6 @@
 #include "fsl_esai.h"
 #include "imx-pcm.h"
 
-#define FSL_ESAI_RATES		SNDRV_PCM_RATE_8000_192000
 #define FSL_ESAI_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
 				SNDRV_PCM_FMTBIT_S16_LE | \
 				SNDRV_PCM_FMTBIT_S20_3LE | \
@@ -647,14 +646,14 @@ static struct snd_soc_dai_driver fsl_esai_dai = {
 		.stream_name = "CPU-Playback",
 		.channels_min = 1,
 		.channels_max = 12,
-		.rates = FSL_ESAI_RATES,
+		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = FSL_ESAI_FORMATS,
 	},
 	.capture = {
 		.stream_name = "CPU-Capture",
 		.channels_min = 1,
 		.channels_max = 8,
-		.rates = FSL_ESAI_RATES,
+		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = FSL_ESAI_FORMATS,
 	},
 	.ops = &fsl_esai_dai_ops,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 9fadf7e31c5f..18e5ce81527d 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -668,7 +668,7 @@ static struct snd_soc_dai_driver fsl_sai_dai = {
 	.playback = {
 		.stream_name = "CPU-Playback",
 		.channels_min = 1,
-		.channels_max = 2,
+		.channels_max = 32,
 		.rate_min = 8000,
 		.rate_max = 192000,
 		.rates = SNDRV_PCM_RATE_KNOT,
@@ -677,7 +677,7 @@ static struct snd_soc_dai_driver fsl_sai_dai = {
 	.capture = {
 		.stream_name = "CPU-Capture",
 		.channels_min = 1,
-		.channels_max = 2,
+		.channels_max = 32,
 		.rate_min = 8000,
 		.rate_max = 192000,
 		.rates = SNDRV_PCM_RATE_KNOT,
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 50349437d961..173cb8496641 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -35,6 +35,7 @@
 #include <linux/module.h>
 #include <linux/interrupt.h>
 #include <linux/clk.h>
+#include <linux/ctype.h>
 #include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
@@ -55,16 +56,6 @@
 #include "imx-pcm.h"
 
 /**
- * FSLSSI_I2S_RATES: sample rates supported by the I2S
- *
- * This driver currently only supports the SSI running in I2S slave mode,
- * which means the codec determines the sample rate.  Therefore, we tell
- * ALSA that we support all rates and let the codec driver decide what rates
- * are really supported.
- */
-#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS
-
-/**
  * FSLSSI_I2S_FORMATS: audio formats supported by the SSI
  *
  * The SSI has a limitation in that the samples must be in the same byte
@@ -224,6 +215,12 @@ struct fsl_ssi_soc_data {
  * @dbg_stats: Debugging statistics
  *
  * @soc: SoC specific data
+ *
+ * @fifo_watermark: the FIFO watermark setting.  Notifies DMA when
+ *             there are @fifo_watermark or fewer words in TX fifo or
+ *             @fifo_watermark or more empty words in RX fifo.
+ * @dma_maxburst: max number of words to transfer in one go.  So far,
+ *             this is always the same as fifo_watermark.
  */
 struct fsl_ssi_private {
 	struct regmap *regs;
@@ -263,6 +260,9 @@ struct fsl_ssi_private {
 
 	const struct fsl_ssi_soc_data *soc;
 	struct device *dev;
+
+	u32 fifo_watermark;
+	u32 dma_maxburst;
 };
 
 /*
@@ -1051,21 +1051,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 	regmap_write(regs, CCSR_SSI_SRCR, srcr);
 	regmap_write(regs, CCSR_SSI_SCR, scr);
 
-	/*
-	 * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't
-	 * use FIFO 1. We program the transmit water to signal a DMA transfer
-	 * if there are only two (or fewer) elements left in the FIFO. Two
-	 * elements equals one frame (left channel, right channel). This value,
-	 * however, depends on the depth of the transmit buffer.
-	 *
-	 * We set the watermark on the same level as the DMA burstsize.  For
-	 * fiq it is probably better to use the biggest possible watermark
-	 * size.
-	 */
-	if (ssi_private->use_dma)
-		wm = ssi_private->fifo_depth - 2;
-	else
-		wm = ssi_private->fifo_depth;
+	wm = ssi_private->fifo_watermark;
 
 	regmap_write(regs, CCSR_SSI_SFCSR,
 			CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) |
@@ -1217,14 +1203,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
 		.stream_name = "CPU-Playback",
 		.channels_min = 1,
 		.channels_max = 32,
-		.rates = FSLSSI_I2S_RATES,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.capture = {
 		.stream_name = "CPU-Capture",
 		.channels_min = 1,
 		.channels_max = 32,
-		.rates = FSLSSI_I2S_RATES,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.ops = &fsl_ssi_dai_ops,
@@ -1330,14 +1316,10 @@ static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = {
  */
 static void make_lowercase(char *s)
 {
-	char *p = s;
-	char c;
-
-	while ((c = *p)) {
-		if ((c >= 'A') && (c <= 'Z'))
-			*p = c + ('a' - 'A');
-		p++;
-	}
+	if (!s)
+		return;
+	for (; *s; s++)
+		*s = tolower(*s);
 }
 
 static int fsl_ssi_imx_probe(struct platform_device *pdev,
@@ -1373,12 +1355,8 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 		dev_dbg(&pdev->dev, "could not get baud clock: %ld\n",
 			 PTR_ERR(ssi_private->baudclk));
 
-	/*
-	 * We have burstsize be "fifo_depth - 2" to match the SSI
-	 * watermark setting in fsl_ssi_startup().
-	 */
-	ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2;
-	ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2;
+	ssi_private->dma_params_tx.maxburst = ssi_private->dma_maxburst;
+	ssi_private->dma_params_rx.maxburst = ssi_private->dma_maxburst;
 	ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0;
 	ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0;
 
@@ -1543,6 +1521,47 @@ static int fsl_ssi_probe(struct platform_device *pdev)
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		ssi_private->fifo_depth = 8;
 
+	/*
+	 * Set the watermark for transmit FIFO 0 and receive FIFO 0. We don't
+	 * use FIFO 1 but set the watermark appropriately nontheless.
+	 * We program the transmit water to signal a DMA transfer
+	 * if there are N elements left in the FIFO. For chips with 15-deep
+	 * FIFOs, set watermark to 8.  This allows the SSI to operate at a
+	 * high data rate without channel slipping. Behavior is unchanged
+	 * for the older chips with a fifo depth of only 8.  A value of 4
+	 * might be appropriate for the older chips, but is left at
+	 * fifo_depth-2 until sombody has a chance to test.
+	 *
+	 * We set the watermark on the same level as the DMA burstsize.  For
+	 * fiq it is probably better to use the biggest possible watermark
+	 * size.
+	 */
+	switch (ssi_private->fifo_depth) {
+	case 15:
+		/*
+		 * 2 samples is not enough when running at high data
+		 * rates (like 48kHz @ 16 bits/channel, 16 channels)
+		 * 8 seems to split things evenly and leave enough time
+		 * for the DMA to fill the FIFO before it's over/under
+		 * run.
+		 */
+		ssi_private->fifo_watermark = 8;
+		ssi_private->dma_maxburst = 8;
+		break;
+	case 8:
+	default:
+		/*
+		 * maintain old behavior for older chips.
+		 * Keeping it the same because I don't have an older
+		 * board to test with.
+		 * I suspect this could be changed to be something to
+		 * leave some more space in the fifo.
+		 */
+		ssi_private->fifo_watermark = ssi_private->fifo_depth - 2;
+		ssi_private->dma_maxburst = ssi_private->fifo_depth - 2;
+		break;
+	}
+
 	dev_set_drvdata(&pdev->dev, ssi_private);
 
 	if (ssi_private->soc->imx) {
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index bb0459018b45..9d19b808f634 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -48,7 +48,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
 	return snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
 }
 
-static struct snd_soc_ops imx_mc13783_hifi_ops = {
+static const struct snd_soc_ops imx_mc13783_hifi_ops = {
 	.hw_params = imx_mc13783_hifi_hw_params,
 };
 
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index f3d3d1ffa84e..314814ddd2b0 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -33,48 +33,20 @@ static bool filter(struct dma_chan *chan, void *param)
 	return true;
 }
 
-static const struct snd_pcm_hardware imx_pcm_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,
-	.buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE,
-	.period_bytes_min = 128,
-	.period_bytes_max = 65535, /* Limited by SDMA engine */
-	.periods_min = 2,
-	.periods_max = 255,
-	.fifo_size = 0,
-};
-
 static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = {
-	.pcm_hardware = &imx_pcm_hardware,
 	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
 	.compat_filter_fn = filter,
-	.prealloc_buffer_size = IMX_DEFAULT_DMABUF_SIZE,
 };
 
 int imx_pcm_dma_init(struct platform_device *pdev, size_t size)
 {
 	struct snd_dmaengine_pcm_config *config;
-	struct snd_pcm_hardware *pcm_hardware;
 
 	config = devm_kzalloc(&pdev->dev,
 			sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL);
 	if (!config)
 		return -ENOMEM;
 	*config = imx_dmaengine_pcm_config;
-	if (size)
-		config->prealloc_buffer_size = size;
-
-	pcm_hardware = devm_kzalloc(&pdev->dev,
-			sizeof(struct snd_pcm_hardware), GFP_KERNEL);
-	*pcm_hardware = imx_pcm_hardware;
-	if (size)
-		pcm_hardware->buffer_bytes_max = size;
-
-	config->pcm_hardware = pcm_hardware;
 
 	return devm_snd_dmaengine_pcm_register(&pdev->dev,
 		config,
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index dac6688540dc..92410f7ca1fa 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -282,7 +282,7 @@ static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int ssi_irq = 0;
+static int ssi_irq;
 
 static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
 {
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 1b60958e2080..206b898e554c 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -33,14 +33,14 @@ struct imx_wm8962_data {
 	struct snd_soc_card card;
 	char codec_dai_name[DAI_NAME_SIZE];
 	char platform_name[DAI_NAME_SIZE];
-	struct clk *codec_clk;
 	unsigned int clk_frequency;
 };
 
 struct imx_priv {
 	struct platform_device *pdev;
+	int sample_rate;
+	snd_pcm_format_t sample_format;
 };
-static struct imx_priv card_priv;
 
 static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
@@ -49,14 +49,14 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
 	SND_SOC_DAPM_MIC("DMIC", NULL),
 };
 
-static int sample_rate = 44100;
-static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE;
-
 static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params)
 {
-	sample_rate = params_rate(params);
-	sample_format = params_format(params);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+
+	priv->sample_rate = params_rate(params);
+	priv->sample_format = params_format(params);
 
 	return 0;
 }
@@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai *codec_dai;
-	struct imx_priv *priv = &card_priv;
+	struct imx_priv *priv = snd_soc_card_get_drvdata(card);
 	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
 	struct device *dev = &priv->pdev->dev;
 	unsigned int pll_out;
@@ -85,10 +85,10 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
 		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
-			if (sample_format == SNDRV_PCM_FORMAT_S24_LE)
-				pll_out = sample_rate * 384;
+			if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+				pll_out = priv->sample_rate * 384;
 			else
-				pll_out = sample_rate * 256;
+				pll_out = priv->sample_rate * 256;
 
 			ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
 					WM8962_FLL_MCLK, data->clk_frequency,
@@ -140,7 +140,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai *codec_dai;
-	struct imx_priv *priv = &card_priv;
+	struct imx_priv *priv = snd_soc_card_get_drvdata(card);
 	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
 	struct device *dev = &priv->pdev->dev;
 	int ret;
@@ -160,13 +160,20 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *ssi_np, *codec_np;
 	struct platform_device *ssi_pdev;
-	struct imx_priv *priv = &card_priv;
 	struct i2c_client *codec_dev;
 	struct imx_wm8962_data *data;
+	struct imx_priv *priv;
+	struct clk *codec_clk;
 	int int_port, ext_port;
 	int ret;
 
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
 	priv->pdev = pdev;
+	priv->sample_rate = 44100;
+	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
 
 	ret = of_property_read_u32(np, "mux-int-port", &int_port);
 	if (ret) {
@@ -231,19 +238,15 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 		goto fail;
 	}
 
-	data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
-	if (IS_ERR(data->codec_clk)) {
-		ret = PTR_ERR(data->codec_clk);
+	codec_clk = clk_get(&codec_dev->dev, NULL);
+	if (IS_ERR(codec_clk)) {
+		ret = PTR_ERR(codec_clk);
 		dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
 		goto fail;
 	}
 
-	data->clk_frequency = clk_get_rate(data->codec_clk);
-	ret = clk_prepare_enable(data->codec_clk);
-	if (ret) {
-		dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret);
-		goto fail;
-	}
+	data->clk_frequency = clk_get_rate(codec_clk);
+	clk_put(codec_clk);
 
 	data->dai.name = "HiFi";
 	data->dai.stream_name = "HiFi";
@@ -258,10 +261,10 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	data->card.dev = &pdev->dev;
 	ret = snd_soc_of_parse_card_name(&data->card, "model");
 	if (ret)
-		goto clk_fail;
+		goto fail;
 	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
 	if (ret)
-		goto clk_fail;
+		goto fail;
 	data->card.num_links = 1;
 	data->card.owner = THIS_MODULE;
 	data->card.dai_link = &data->dai;
@@ -277,16 +280,9 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
-		goto clk_fail;
+		goto fail;
 	}
 
-	of_node_put(ssi_np);
-	of_node_put(codec_np);
-
-	return 0;
-
-clk_fail:
-	clk_disable_unprepare(data->codec_clk);
 fail:
 	of_node_put(ssi_np);
 	of_node_put(codec_np);
@@ -294,17 +290,6 @@ fail:
 	return ret;
 }
 
-static int imx_wm8962_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
-
-	if (!IS_ERR(data->codec_clk))
-		clk_disable_unprepare(data->codec_clk);
-
-	return 0;
-}
-
 static const struct of_device_id imx_wm8962_dt_ids[] = {
 	{ .compatible = "fsl,imx-audio-wm8962", },
 	{ /* sentinel */ }
@@ -318,7 +303,6 @@ static struct platform_driver imx_wm8962_driver = {
 		.of_match_table = imx_wm8962_dt_ids,
 	},
 	.probe = imx_wm8962_probe,
-	.remove = imx_wm8962_remove,
 };
 module_platform_driver(imx_wm8962_driver);
 
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 243700cc29e6..07ee355ee385 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -25,7 +25,6 @@
 #include <asm/mpc52xx_psc.h>
 
 #include "mpc5200_dma.h"
-#include "mpc5200_psc_ac97.h"
 
 #define DRV_NAME "mpc5200-psc-ac97"
 
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h
deleted file mode 100644
index e881e784b270..000000000000
--- a/sound/soc/fsl/mpc5200_psc_ac97.h
+++ /dev/null
@@ -1,13 +0,0 @@
-/*
- * Freescale MPC5200 PSC in AC97 mode
- * ALSA SoC Digital Audio Interface (DAI) driver
- *
- */
-
-#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
-#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
-
-#define MPC5200_AC97_NORMAL 0
-#define MPC5200_AC97_SPDIF 1
-
-#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index ddf49f30b23f..a639b52c16f6 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -174,7 +174,7 @@ static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card)
 /**
  * mpc8610_hpcd_ops: ASoC machine driver operations
  */
-static struct snd_soc_ops mpc8610_hpcd_ops = {
+static const struct snd_soc_ops mpc8610_hpcd_ops = {
 	.startup = mpc8610_hpcd_startup,
 };
 
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index 198eeb3f3f7a..d7ec3d20065c 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -73,7 +73,7 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops mx27vis_aic32x4_snd_ops = {
+static const struct snd_soc_ops mx27vis_aic32x4_snd_ops = {
 	.hw_params	= mx27vis_aic32x4_hw_params,
 };
 
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index a1f780ecadf5..41c623c55c16 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -184,7 +184,7 @@ static int p1022_ds_machine_remove(struct snd_soc_card *card)
 /**
  * p1022_ds_ops: ASoC machine driver operations
  */
-static struct snd_soc_ops p1022_ds_ops = {
+static const struct snd_soc_ops p1022_ds_ops = {
 	.startup = p1022_ds_startup,
 };
 
diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c
index d4d88a8cb9c0..4afbdd610bfa 100644
--- a/sound/soc/fsl/p1022_rdk.c
+++ b/sound/soc/fsl/p1022_rdk.c
@@ -188,7 +188,7 @@ static int p1022_rdk_machine_remove(struct snd_soc_card *card)
 /**
  * p1022_rdk_ops: ASoC machine driver operations
  */
-static struct snd_soc_ops p1022_rdk_ops = {
+static const struct snd_soc_ops p1022_rdk_ops = {
 	.startup = p1022_rdk_startup,
 };
 
diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c
index ae403c29688f..66fb6c4614d2 100644
--- a/sound/soc/fsl/phycore-ac97.c
+++ b/sound/soc/fsl/phycore-ac97.c
@@ -23,7 +23,7 @@
 
 static struct snd_soc_card imx_phycore;
 
-static struct snd_soc_ops imx_phycore_hifi_ops = {
+static const struct snd_soc_ops imx_phycore_hifi_ops = {
 };
 
 static struct snd_soc_dai_link imx_phycore_dai_ac97[] = {
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index b454972dce35..cdaf16367b47 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -139,7 +139,7 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops wm1133_ev1_ops = {
+static const struct snd_soc_ops wm1133_ev1_ops = {
 	.hw_params = wm1133_ev1_hw_params,
 };