Merge commit 'v2.6.30-rc6' into perfcounters/core

Merge reason: this branch was on an -rc4 base, merge it up to -rc6
              to get the latest upstream fixes.

Signed-off-by: Ingo Molnar <mingo@elte.hu>

11 files changed, 157 insertions, 71 deletions

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a259148..594c6c5b7838 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
 	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-			    SNDRV_PCM_INFO_INTERLEAVED,
+			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
 	.formats	  = AU1XPSC_PCM_FMTS,
 	.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
 	.period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..df7c8c281d2f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		4, 3, 0, output_tvl),
 
 	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+		TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
 	/* Common capture gain controls */
 	SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed9..0275321ff8ab 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
 	 * required for LRC in master mode. The DACs or ADCs need a
 	 * valid audio path i.e. pin -> ADC or DAC -> pin before
 	 * the LRC will be enabled in master mode. */
-	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+	if (!master || cmd != SNDRV_PCM_TRIGGER_START)
 		return 0;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3f..40cd274eb1ef 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
 	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
 	&wm8990_dapm_ainlmux_controls, inmixer_event,
 	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
 	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
 	&wm8990_dapm_ainrmux_controls, inmixer_event,
 	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
 	/* LIN34 PGA */
 	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
-	{"LIN34 PGA", "LIN4 Switch", "LIN4"},
+	{"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
 	/* INMIXL */
 	{"INMIXL", "Record Left Volume", "LOMIX"},
 	{"INMIXL", "LIN2 Volume", "LIN2"},
 	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
 	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
-	/* AILNMUX */
-	{"AILNMUX", "INMIXL Mix", "INMIXL"},
-	{"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
-	{"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
-	{"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
-	{"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* AINLMUX */
+	{"AINLMUX", "INMIXL Mix", "INMIXL"},
+	{"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+	{"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+	{"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
 	/* ADC */
-	{"Left ADC", NULL, "AILNMUX"},
+	{"Left ADC", NULL, "AINLMUX"},
 
 	/* RIN12 PGA */
 	{"RIN12 PGA", "RIN1 Switch", "RIN1"},
 	{"RIN12 PGA", "RIN2 Switch", "RIN2"},
 	/* RIN34 PGA */
 	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
-	{"RIN34 PGA", "RIN4 Switch", "RIN4"},
+	{"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
 	/* INMIXL */
 	{"INMIXR", "Record Right Volume", "ROMIX"},
 	{"INMIXR", "RIN2 Volume", "RIN2"},
 	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
 	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
-	/* AIRNMUX */
-	{"AIRNMUX", "INMIXR Mix", "INMIXR"},
-	{"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
-	{"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
-	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
-	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* AINRMUX */
+	{"AINRMUX", "INMIXR Mix", "INMIXR"},
+	{"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+	{"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+	{"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
 	/* ADC */
-	{"Right ADC", NULL, "AIRNMUX"},
+	{"Right ADC", NULL, "AINRMUX"},
 
 	/* LOMIX */
 	{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
 
 	/* OUT3MIX */
-	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
 	{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
 
 	/* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	/* Output Pins */
 	{"LON", NULL, "LONMIX"},
 	{"LOP", NULL, "LOPMIX"},
-	{"OUT", NULL, "OUT3MIX"},
+	{"OUT3", NULL, "OUT3MIX"},
 	{"LOUT", NULL, "LOUT PGA"},
 	{"SPKN", NULL, "SPKMIX"},
 	{"ROUT", NULL, "ROUT PGA"},
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c9657e..411a710be660 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
 	tristate
 
 config SND_DAVINCI_SOC_EVM
-	tristate "SoC Audio support for DaVinci EVM"
-	depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+	tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+	depends on SND_DAVINCI_SOC
+	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
 	select SND_DAVINCI_SOC_I2S
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on TI
-	  DaVinci EVM platform.
+	  DaVinci DM6446 or DM355 EVM platforms.
 
 config SND_DAVINCI_SOC_SFFSDR
 	tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b347007c..58fd1cbedd88 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
 
 /* evm audio private data */
 static struct aic3x_setup_data evm_aic3x_setup = {
-	.i2c_bus = 0,
+	.i2c_bus = 1,
 	.i2c_address = 0x1b,
 };
 
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
 	.codec_data = &evm_aic3x_setup,
 };
 
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
 static struct resource evm_snd_resources[] = {
 	{
-		.start = DAVINCI_MCBSP_BASE,
-		.end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+		.start = DAVINCI_ASP0_BASE,
+		.end = DAVINCI_ASP0_BASE + SZ_8K - 1,
 		.flags = IORESOURCE_MEM,
 	},
 };
 
 static struct evm_snd_platform_data evm_snd_data = {
-	.tx_dma_ch	= DM644X_DMACH_MCBSP_TX,
-	.rx_dma_ch	= DM644X_DMACH_MCBSP_RX,
+	.tx_dma_ch	= DAVINCI_DMA_ASP0_TX,
+	.rx_dma_ch	= DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+	{
+		.start = DAVINCI_ASP1_BASE,
+		.end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+		.flags = IORESOURCE_MEM,
+	},
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+	.tx_dma_ch	= DAVINCI_DMA_ASP1_TX,
+	.rx_dma_ch	= DAVINCI_DMA_ASP1_RX,
 };
 
 static struct platform_device *evm_snd_device;
 
 static int __init evm_init(void)
 {
+	struct resource *resources;
+	unsigned num_resources;
+	struct evm_snd_platform_data *data;
+	int index;
 	int ret;
 
-	evm_snd_device = platform_device_alloc("soc-audio", 0);
+	if (machine_is_davinci_evm()) {
+		davinci_cfg_reg(DM644X_MCBSP);
+
+		resources = evm_snd_resources;
+		num_resources = ARRAY_SIZE(evm_snd_resources);
+		data = &evm_snd_data;
+		index = 0;
+	} else if (machine_is_davinci_dm355_evm()) {
+		/* we don't use ASP1 IRQs, or we'd need to mux them ... */
+		davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+		davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+		resources = dm335evm_snd_resources;
+		num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+		data = &dm335evm_snd_data;
+		index = 1;
+	} else
+		return -EINVAL;
+
+	evm_snd_device = platform_device_alloc("soc-audio", index);
 	if (!evm_snd_device)
 		return -ENOMEM;
 
 	platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
 	evm_snd_devdata.dev = &evm_snd_device->dev;
-	platform_device_add_data(evm_snd_device, &evm_snd_data,
-				 sizeof(evm_snd_data));
+	platform_device_add_data(evm_snd_device, data, sizeof(*data));
 
-	ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
-					    ARRAY_SIZE(evm_snd_resources));
+	ret = platform_device_add_resources(evm_snd_device, resources,
+			num_resources);
 	if (ret) {
 		platform_device_put(evm_snd_device);
 		return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb9439d3d8..b1ea52fc83c7 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
 
 #include "davinci-pcm.h"
 
+
+/*
+ * NOTE:  terminology here is confusing.
+ *
+ *  - This driver supports the "Audio Serial Port" (ASP),
+ *    found on dm6446, dm355, and other DaVinci chips.
+ *
+ *  - But it labels it a "Multi-channel Buffered Serial Port"
+ *    (McBSP) as on older chips like the dm642 ... which was
+ *    backward-compatible, possibly explaining that confusion.
+ *
+ *  - OMAP chips have a controller called McBSP, which is
+ *    incompatible with the DaVinci flavor of McBSP.
+ *
+ *  - Newer DaVinci chips have a controller called McASP,
+ *    incompatible with ASP and with either McBSP.
+ *
+ * In short:  this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
 #define DAVINCI_MCBSP_DRR_REG	0x00
 #define DAVINCI_MCBSP_DXR_REG	0x04
 #define DAVINCI_MCBSP_SPCR_REG	0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *ioarea;
 	struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 
 	cpu_dai->private_data = dev;
 
-	dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+	dev->clk = clk_get(&pdev->dev, NULL);
 	if (IS_ERR(dev->clk)) {
 		ret = -ENODEV;
 		goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
 	struct resource *mem;
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b3a53d..a05996588489 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
+#include <mach/edma.h>
 
 #include "davinci-pcm.h"
 
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
 	spinlock_t lock;
 	int period;		/* current DMA period */
 	int master_lch;		/* Master DMA channel */
-	int slave_lch;		/* Slave DMA channel */
+	int slave_lch;		/* linked parameter RAM reload slot */
 	struct davinci_pcm_dma_params *params;	/* DMA params */
 };
 
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 		dst_bidx = data_type;
 	}
 
-	davinci_set_dma_src_params(lch, src, INCR, W8BIT);
-	davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
-	davinci_set_dma_src_index(lch, src_bidx, 0);
-	davinci_set_dma_dest_index(lch, dst_bidx, 0);
-	davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+	edma_set_src(lch, src, INCR, W8BIT);
+	edma_set_dest(lch, dst, INCR, W8BIT);
+	edma_set_src_index(lch, src_bidx, 0);
+	edma_set_dest_index(lch, dst_bidx, 0);
+	edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
 
 	prtd->period++;
 	if (unlikely(prtd->period >= runtime->periods))
 		prtd->period = 0;
 }
 
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 {
 	struct snd_pcm_substream *substream = data;
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
-	int tcc = TCC_ANY;
+	struct edmacc_param p_ram;
 	int ret;
 
 	if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	prtd->params = dma_data;
 
 	/* Request master DMA channel */
-	ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+	ret = edma_alloc_channel(prtd->params->channel,
 				  davinci_pcm_dma_irq, substream,
-				  &prtd->master_lch, &tcc, EVENTQ_0);
-	if (ret)
+				  EVENTQ_0);
+	if (ret < 0)
 		return ret;
+	prtd->master_lch = ret;
 
-	/* Request slave DMA channel */
-	ret = davinci_request_dma(PARAM_ANY, "Link",
-				  NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
-	if (ret) {
-		davinci_free_dma(prtd->master_lch);
+	/* Request parameter RAM reload slot */
+	ret = edma_alloc_slot(EDMA_SLOT_ANY);
+	if (ret < 0) {
+		edma_free_channel(prtd->master_lch);
 		return ret;
 	}
-
-	/* Link slave DMA channel in loopback */
-	davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+	prtd->slave_lch = ret;
+
+	/* Issue transfer completion IRQ when the channel completes a
+	 * transfer, then always reload from the same slot (by a kind
+	 * of loopback link).  The completion IRQ handler will update
+	 * the reload slot with a new buffer.
+	 *
+	 * REVISIT save p_ram here after setting up everything except
+	 * the buffer and its length (ccnt) ... use it as a template
+	 * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+	 */
+	edma_read_slot(prtd->slave_lch, &p_ram);
+	p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+	p_ram.link_bcntrld = prtd->slave_lch << 5;
+	edma_write_slot(prtd->slave_lch, &p_ram);
 
 	return 0;
 }
@@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		davinci_start_dma(prtd->master_lch);
+		edma_start(prtd->master_lch);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		davinci_stop_dma(prtd->master_lch);
+		edma_stop(prtd->master_lch);
 		break;
 	default:
 		ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct paramentry_descriptor temp;
+	struct edmacc_param temp;
 
 	prtd->period = 0;
 	davinci_pcm_enqueue_dma(substream);
 
-	/* Get slave channel dma params for master channel startup */
-	davinci_get_dma_params(prtd->slave_lch, &temp);
-	davinci_set_dma_params(prtd->master_lch, &temp);
+	/* Copy self-linked parameter RAM entry into master channel */
+	edma_read_slot(prtd->slave_lch, &temp);
+	edma_write_slot(prtd->master_lch, &temp);
 
 	return 0;
 }
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
 
 	spin_lock(&prtd->lock);
 
-	davinci_dma_getposition(prtd->master_lch, &src, &dst);
+	edma_get_position(prtd->master_lch, &src, &dst);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		count = src - runtime->dma_addr;
 	else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct davinci_runtime_data *prtd = runtime->private_data;
 
-	davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+	edma_unlink(prtd->slave_lch);
 
-	davinci_free_dma(prtd->slave_lch);
-	davinci_free_dma(prtd->master_lch);
+	edma_free_slot(prtd->slave_lch);
+	edma_free_channel(prtd->master_lch);
 
 	kfree(prtd);
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..1111c710118a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 
 static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
 		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
 	.rate_min = 8000,
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP_VALID),
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BATCH),
 	.formats =	DMABRG_FMTS,
 	.rates =	DMABRG_RATES,
 	.rate_min =		8000,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f652d0d..1cd149b9ce69 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
+	if (!card->instantiated)
+		return 0;
+
 	run_delayed_work(&card->delayed_work);
 
 	if (platform->remove)