Merge branch 'for-2.6.31' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc

12 files changed, 330 insertions, 64 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index b642c21d2f65..1ae5a81f4436 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5286,7 +5286,7 @@ P:	Liam Girdwood
 M:	lrg@slimlogic.co.uk
 P:	Mark Brown
 M:	broonie@opensource.wolfsonmicro.com
-T:	git git://opensource.wolfsonmicro.com/linux-2.6-asoc
+T:	git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
 L:	alsa-devel@alsa-project.org (subscribers-only)
 W:	http://alsa-project.org/main/index.php/ASoC
 S:	Supported
diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h
new file mode 100644
index 000000000000..e0abc437d796
--- /dev/null
+++ b/arch/arm/mach-davinci/include/mach/asp.h
@@ -0,0 +1,25 @@
+/*
+ * <mach/asp.h> - DaVinci Audio Serial Port support
+ */
+#ifndef __ASM_ARCH_DAVINCI_ASP_H
+#define __ASM_ARCH_DAVINCI_ASP_H
+
+#include <mach/irqs.h>
+
+/* Bases of register banks */
+#define DAVINCI_ASP0_BASE	0x01E02000
+#define DAVINCI_ASP1_BASE	0x01E04000
+
+/* EDMA channels */
+#define DAVINCI_DMA_ASP0_TX	2
+#define DAVINCI_DMA_ASP0_RX	3
+#define DAVINCI_DMA_ASP1_TX	8
+#define DAVINCI_DMA_ASP1_RX	9
+
+/* Interrupts */
+#define DAVINCI_ASP0_RX_INT	IRQ_MBRINT
+#define DAVINCI_ASP0_TX_INT	IRQ_MBXINT
+#define DAVINCI_ASP1_RX_INT	IRQ_MBRINT
+#define DAVINCI_ASP1_TX_INT	IRQ_MBXINT
+
+#endif /* __ASM_ARCH_DAVINCI_ASP_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 496dc30457b7..352d7eee9b6d 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -79,7 +79,8 @@ struct snd_pcm_substream;
 #define SND_SOC_CLOCK_OUT		1
 
 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
-                               SNDRV_PCM_FMTBIT_S32_LE)
+                               SNDRV_PCM_FMTBIT_S32_LE |\
+                               SNDRV_PCM_FMTBIT_S32_BE)
 
 struct snd_soc_dai_ops;
 struct snd_soc_dai;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 550c903f23bf..1fd4e88f50cf 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -585,6 +585,8 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
 	}
 
 	soc_ac97_ops.reset(codec->ac97);
+	if (soc_ac97_ops.warm_reset)
+		soc_ac97_ops.warm_reset(codec->ac97);
 	if (ac97_read(codec, 0) != wm9712_reg[0])
 		goto err;
 	return 0;
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/omap/Kconfig b/sound/soc/omap/Kconfig
index 675732e724d5..b771238662b6 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -39,6 +39,14 @@ config SND_OMAP_SOC_OMAP2EVM
 	help
 	  Say Y if you want to add support for SoC audio on the omap2evm board.
 
+config SND_OMAP_SOC_OMAP3EVM
+	tristate "SoC Audio support for OMAP3EVM board"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the omap3evm board.
+
 config SND_OMAP_SOC_SDP3430
 	tristate "SoC Audio support for Texas Instruments SDP3430"
 	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 0c9e4ac37660..a37f49862389 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -10,6 +10,7 @@ snd-soc-n810-objs := n810.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
@@ -18,6 +19,7 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
new file mode 100644
index 000000000000..9114c263077b
--- /dev/null
+++ b/sound/soc/omap/omap3evm.c
@@ -0,0 +1,147 @@
+/*
+ * omap3evm.c  -- ALSA SoC support for OMAP3 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2008 Texas Instruments, Incorporated
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3evm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap3evm_ops = {
+	.hw_params = omap3evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3evm_dai = {
+	.name 		= "TWL4030",
+	.stream_name 	= "TWL4030",
+	.cpu_dai 	= &omap_mcbsp_dai[0],
+	.codec_dai 	= &twl4030_dai[TWL4030_DAI_HIFI],
+	.ops 		= &omap3evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3evm = {
+	.name = "omap3evm",
+	.platform = &omap_soc_platform,
+	.dai_link = &omap3evm_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3evm_snd_devdata = {
+	.card = &snd_soc_omap3evm,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3evm_snd_device;
+
+static int __init omap3evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3evm()) {
+		pr_err("Not OMAP3 EVM!\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3 EVM SoC init\n");
+
+	omap3evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap3evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata);
+	omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev;
+	*(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1;
+
+	ret = platform_device_add(omap3evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap3evm_snd_device);
+
+	return ret;
+}
+
+static void __exit omap3evm_soc_exit(void)
+{
+	platform_device_unregister(omap3evm_snd_device);
+}
+
+module_init(omap3evm_soc_init);
+module_exit(omap3evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 60145770aeba..4743e262895d 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -106,10 +106,8 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
 	if (IS_ERR(clk_i2s))
 		return PTR_ERR(clk_i2s);
 
-	if (!cpu_dai->active) {
-		SACR0 |= SACR0_RST;
+	if (!cpu_dai->active)
 		SACR0 = 0;
-	}
 
 	return 0;
 }
@@ -178,9 +176,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	/* is port used by another stream */
 	if (!(SACR0 & SACR0_ENB)) {
-
 		SACR0 = 0;
-		SACR1 = 0;
 		if (pxa_i2s.master)
 			SACR0 |= SACR0_BCKD;
 
@@ -226,6 +222,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			SACR1 &= ~SACR1_DRPL;
+		else
+			SACR1 &= ~SACR1_DREC;
 		SACR0 |= SACR0_ENB;
 		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
@@ -252,21 +252,16 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
 		SAIMR &= ~SAIMR_RFS;
 	}
 
-	if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
+	if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
 		SACR0 &= ~SACR0_ENB;
 		pxa_i2s_wait();
 		clk_disable(clk_i2s);
 	}
-
-	clk_put(clk_i2s);
 }
 
 #ifdef CONFIG_PM
 static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
-	if (!dai->active)
-		return 0;
-
 	/* store registers */
 	pxa_i2s.sacr0 = SACR0;
 	pxa_i2s.sacr1 = SACR1;
@@ -281,16 +276,14 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 
 static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
-	if (!dai->active)
-		return 0;
-
 	pxa_i2s_wait();
 
-	SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
+	SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB;
 	SACR1 = pxa_i2s.sacr1;
 	SAIMR = pxa_i2s.saimr;
 	SADIV = pxa_i2s.sadiv;
-	SACR0 |= SACR0_ENB;
+
+	SACR0 = pxa_i2s.sacr0;
 
 	return 0;
 }
@@ -347,6 +340,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev)
 	if (ret != 0)
 		clk_put(clk_i2s);
 
+	/*
+	 * PXA Developer's Manual:
+	 * If SACR0[ENB] is toggled in the middle of a normal operation,
+	 * the SACR0[RST] bit must also be set and cleared to reset all
+	 * I2S controller registers.
+	 */
+	SACR0 = SACR0_RST;
+	SACR0 = 0;
+	/* Make sure RPL and REC are disabled */
+	SACR1 = SACR1_DRPL | SACR1_DREC;
+	/* Along with FIFO servicing */
+	SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
+
 	return ret;
 }