From 1b340bd7e444f20eb2df88c65fa34960c4736ee9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 30 Jul 2008 19:12:04 +0100
Subject: ASoC: Add PXA SSP support

The SSP ports PXA series processors can be used to implement a variety of
audio interface formats. This patch implements support for I2S, DSP A and
DSP B modes on these ports.

This patch is based on the previous out of tree pxa2xx-ssp driver (which
was originally written by Liam Girdwood with updates from Philipp Zabel
and Nicola Perrino) and pxa3xx-ssp driver (originally written by Seth
Forsee based on the pxa2xx-ssp driver). Testing coverage is not complete
currently.

Tested-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Kconfig   |  13 +
 sound/soc/pxa/Makefile  |   2 +
 sound/soc/pxa/pxa-ssp.c | 929 ++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/pxa/pxa-ssp.h |  47 +++
 4 files changed, 991 insertions(+)
 create mode 100644 sound/soc/pxa/pxa-ssp.c
 create mode 100644 sound/soc/pxa/pxa-ssp.h

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f8c1cdd940a..4235524238f 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -21,6 +21,9 @@ config SND_PXA2XX_SOC_AC97
 config SND_PXA2XX_SOC_I2S
 	tristate
 
+config SND_PXA_SOC_SSP
+	tristate
+
 config SND_PXA2XX_SOC_CORGI
 	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
 	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -75,3 +78,13 @@ config SND_PXA2XX_SOC_EM_X270
 	help
 	  Say Y if you want to add support for SoC audio on
 	  CompuLab EM-x270.
+
+config SND_SOC_ZYLONITE
+	tristate "SoC Audio support for Marvell Zylonite"
+	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+	select SND_PXA2XX_SOC_AC97
+	select SND_PXA_SOC_SSP
+	select SND_SOC_WM9713
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Zylonite reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 5bc8edf9dca..00258abb84a 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -2,10 +2,12 @@
 snd-soc-pxa2xx-objs := pxa2xx-pcm.o
 snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
 snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+snd-soc-pxa-ssp-objs := pxa-ssp.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
 obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
 obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
 
 # PXA Machine Support
 snd-soc-corgi-objs := corgi.o
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
new file mode 100644
index 00000000000..e2b54b88c38
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -0,0 +1,929 @@
+#define DEBUG
+/*
+ * pxa-ssp.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright 2005,2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ * TODO:
+ *  o Test network mode for > 16bit sample size
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/regs-ssp.h>
+#include <mach/audio.h>
+#include <mach/ssp.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa-ssp.h"
+
+/*
+ * SSP audio private data
+ */
+struct ssp_priv {
+	struct ssp_dev dev;
+	unsigned int sysclk;
+	int dai_fmt;
+#ifdef CONFIG_PM
+	struct ssp_state state;
+#endif
+};
+
+#define PXA2xx_SSP1_BASE	0x41000000
+#define PXA27x_SSP2_BASE	0x41700000
+#define PXA27x_SSP3_BASE	0x41900000
+#define PXA3xx_SSP4_BASE	0x41a00000
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
+	.name			= "SSP1 PCM Mono out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
+	.name			= "SSP1 PCM Mono in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
+	.name			= "SSP1 PCM Stereo out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
+	.name			= "SSP1 PCM Stereo in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
+	.name			= "SSP2 PCM Mono out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
+	.name			= "SSP2 PCM Mono in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
+	.name			= "SSP2 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
+	.name			= "SSP2 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
+	.name			= "SSP3 PCM Mono out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
+	.name			= "SSP3 PCM Mono in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
+	.name			= "SSP3 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
+	.name			= "SSP3 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
+	.name			= "SSP4 PCM Mono out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
+	.name			= "SSP4 PCM Mono in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
+	.name			= "SSP4 PCM Stereo out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
+	.name			= "SSP4 PCM Stereo in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static void dump_registers(struct ssp_device *ssp)
+{
+	dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
+		 ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1),
+		 ssp_read_reg(ssp, SSTO));
+
+	dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
+		 ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR),
+		 ssp_read_reg(ssp, SSACD));
+}
+
+static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
+	{
+		&pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
+		&pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
+		&pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
+		&pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
+		&pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
+	},
+};
+
+static int pxa_ssp_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	int ret = 0;
+
+	if (!cpu_dai->active) {
+		ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
+		if (ret < 0)
+			return ret;
+		ssp_disable(&priv->dev);
+	}
+	return ret;
+}
+
+static void pxa_ssp_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active) {
+		ssp_disable(&priv->dev);
+		ssp_exit(&priv->dev);
+	}
+}
+
+#ifdef CONFIG_PM
+
+static int pxa_ssp_suspend(struct platform_device *pdev,
+	struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	ssp_save_state(&priv->dev, &priv->state);
+	clk_disable(priv->dev.ssp->clk);
+	return 0;
+}
+
+static int pxa_ssp_resume(struct platform_device *pdev,
+	struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	clk_enable(priv->dev.ssp->clk);
+	ssp_restore_state(&priv->dev, &priv->state);
+	ssp_enable(&priv->dev);
+
+	return 0;
+}
+
+#else
+#define pxa_ssp_suspend	NULL
+#define pxa_ssp_resume	NULL
+#endif
+
+/**
+ * ssp_set_clkdiv - set SSP clock divider
+ * @div: serial clock rate divider
+ */
+static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+{
+	struct ssp_device *ssp = dev->ssp;
+	u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+
+	ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+	int clk_id, unsigned int freq, int dir)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+
+	dev_dbg(&ssp->pdev->dev,
+		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
+		cpu_dai->id, clk_id, freq);
+
+	switch (clk_id) {
+	case PXA_SSP_CLK_NET_PLL:
+		sscr0 |= SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_PLL:
+		/* Internal PLL is fixed */
+		if (cpu_is_pxa25x())
+			priv->sysclk = 1843200;
+		else
+			priv->sysclk = 13000000;
+		break;
+	case PXA_SSP_CLK_EXT:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_ECS;
+		break;
+	case PXA_SSP_CLK_NET:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_NCS | SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_AUDIO:
+		priv->sysclk = 0;
+		ssp_set_scr(&priv->dev, 1);
+		sscr0 |= SSCR0_ADC;
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	/* The SSP clock must be disabled when changing SSP clock mode
+	 * on PXA2xx.  On PXA3xx it must be enabled when doing so. */
+	if (!cpu_is_pxa3xx())
+		clk_disable(priv->dev.ssp->clk);
+	val = ssp_read_reg(ssp, SSCR0) | sscr0;
+	ssp_write_reg(ssp, SSCR0, val);
+	if (!cpu_is_pxa3xx())
+		clk_enable(priv->dev.ssp->clk);
+
+	return 0;
+}
+
+/*
+ * Set the SSP clock dividers.
+ */
+static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+	int div_id, int div)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (div_id) {
+	case PXA_SSP_AUDIO_DIV_ACDS:
+		val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_AUDIO_DIV_SCDB:
+		val = ssp_read_reg(ssp, SSACD);
+		val &= ~SSACD_SCDB;
+#if defined(CONFIG_PXA3xx)
+		if (cpu_is_pxa3xx())
+			val &= ~SSACD_SCDX8;
+#endif
+		switch (div) {
+		case PXA_SSP_CLK_SCDB_1:
+			val |= SSACD_SCDB;
+			break;
+		case PXA_SSP_CLK_SCDB_4:
+			break;
+#if defined(CONFIG_PXA3xx)
+		case PXA_SSP_CLK_SCDB_8:
+			if (cpu_is_pxa3xx())
+				val |= SSACD_SCDX8;
+			else
+				return -EINVAL;
+			break;
+#endif
+		default:
+			return -EINVAL;
+		}
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_DIV_SCR:
+		ssp_set_scr(&priv->dev, div);
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+/*
+ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
+ */
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
+	int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70;
+
+#if defined(CONFIG_PXA3xx)
+	if (cpu_is_pxa3xx())
+		ssp_write_reg(ssp, SSACDD, 0);
+#endif
+
+	switch (freq_out) {
+	case 5622000:
+		break;
+	case 11345000:
+		ssacd |= (0x1 << 4);
+		break;
+	case 12235000:
+		ssacd |= (0x2 << 4);
+		break;
+	case 14857000:
+		ssacd |= (0x3 << 4);
+		break;
+	case 32842000:
+		ssacd |= (0x4 << 4);
+		break;
+	case 48000000:
+		ssacd |= (0x5 << 4);
+		break;
+	case 0:
+		/* Disable */
+		break;
+
+	default:
+#ifdef CONFIG_PXA3xx
+		/* PXA3xx has a clock ditherer which can be used to generate
+		 * a wider range of frequencies - calculate a value for it.
+		 */
+		if (cpu_is_pxa3xx()) {
+			u32 val;
+			u64 tmp = 19968;
+			tmp *= 1000000;
+			do_div(tmp, freq_out);
+			val = tmp;
+
+			val = (val << 16) | 64;;
+			ssp_write_reg(ssp, SSACDD, val);
+
+			ssacd |= (0x6 << 4);
+
+			dev_dbg(&ssp->pdev->dev,
+				"Using SSACDD %x to supply %dHz\n",
+				val, freq_out);
+			break;
+		}
+#endif
+
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSACD, ssacd);
+
+	return 0;
+}
+
+/*
+ * Set the active slots in TDM/Network mode
+ */
+static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+	unsigned int mask, int slots)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7);
+
+	/* set number of active slots */
+	sscr0 |= SSCR0_SlotsPerFrm(slots);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* set active slot mask */
+	ssp_write_reg(ssp, SSTSA, mask);
+	ssp_write_reg(ssp, SSRSA, mask);
+	return 0;
+}
+
+/*
+ * Tristate the SSP DAI lines
+ */
+static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
+	int tristate)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr1;
+
+	sscr1 = ssp_read_reg(ssp, SSCR1);
+	if (tristate)
+		sscr1 &= ~SSCR1_TTE;
+	else
+		sscr1 |= SSCR1_TTE;
+	ssp_write_reg(ssp, SSCR1, sscr1);
+
+	return 0;
+}
+
+/*
+ * Set up the SSP DAI format.
+ * The SSP Port must be inactive before calling this function as the
+ * physical interface format is changed.
+ */
+static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+	u32 sscr1;
+	u32 sspsp;
+
+	/* reset port settings */
+	sscr0 = ssp_read_reg(ssp, SSCR0) &
+		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
+	sspsp = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		sscr1 |= SSCR1_SCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
+		sspsp |= SSPSP_FSRT;
+	case SND_SOC_DAIFMT_DSP_B:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	dump_registers(ssp);
+
+	/* Since we are configuring the timings for the format by hand
+	 * we have to defer some things until hw_params() where we
+	 * know parameters like the sample size.
+	 */
+	priv->dai_fmt = fmt;
+
+	return 0;
+}
+
+/*
+ * Set the SSP audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int pxa_ssp_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 *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int dma = 0, chn = params_channels(params);
+	u32 sscr0;
+	u32 sspsp;
+	int width = snd_pcm_format_physical_width(params_format(params));
+
+	/* select correct DMA params */
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		dma = 1; /* capture DMA offset is 1,3 */
+	if (chn == 2)
+		dma += 2; /* stereo DMA offset is 2, mono is 0 */
+	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
+
+	dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+
+	/* we can only change the settings if the port is not in use */
+	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+		return 0;
+
+	/* clear selected SSP bits */
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* bit size */
+	sscr0 = ssp_read_reg(ssp, SSCR0);
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+#ifdef CONFIG_PXA3xx
+		if (cpu_is_pxa3xx())
+			sscr0 |= SSCR0_FPCKE;
+#endif
+		sscr0 |= SSCR0_DataSize(16);
+		if (params_channels(params) > 1)
+			sscr0 |= SSCR0_EDSS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
+		/* we must be in network mode (2 slots) for 24 bit stereo */
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
+		/* we must be in network mode (2 slots) for 32 bit stereo */
+		break;
+	}
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* Cleared when the DAI format is set */
+		sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+		ssp_write_reg(ssp, SSPSP, sspsp);
+		break;
+	default:
+		break;
+	}
+
+	/* We always use a network mode so we always require TDM slots
+	 * - complain loudly and fail if they've not been set up yet.
+	 */
+	if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
+		return -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return 0;
+}
+
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret = 0;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_RESUME:
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		val = ssp_read_reg(ssp, SSSR);
+		ssp_write_reg(ssp, SSSR, val);
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		ssp_disable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return ret;
+}
+
+static int pxa_ssp_probe(struct platform_device *pdev,
+			    struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv;
+	int ret;
+
+	priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+	if (priv->dev.ssp == NULL) {
+		ret = -ENODEV;
+		goto err_priv;
+	}
+
+	dai->private_data = priv;
+
+	return 0;
+
+err_priv:
+	kfree(priv);
+	return ret;
+}
+
+static void pxa_ssp_remove(struct platform_device *pdev,
+			      struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv = dai->private_data;
+	ssp_free(priv->dev.ssp);
+}
+
+#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+			  SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |	\
+			  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |	\
+			  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			    SNDRV_PCM_FMTBIT_S24_LE |	\
+			    SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai pxa_ssp_dai[] = {
+	{
+		.name = "pxa2xx-ssp1",
+		.id = 0,
+		.type = SND_SOC_DAI_PCM,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			 .channels_min = 1,
+			 .channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+		 },
+		.dai_ops = {
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{	.name = "pxa2xx-ssp2",
+		.id = 1,
+		.type = SND_SOC_DAI_PCM,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+		 },
+		.dai_ops = {
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp3",
+		.id = 2,
+		.type = SND_SOC_DAI_PCM,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+		 },
+		.dai_ops = {
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp4",
+		.id = 3,
+		.type = SND_SOC_DAI_PCM,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+		 },
+		.dai_ops = {
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+};
+EXPORT_SYMBOL_GPL(pxa_ssp_dai);
+
+/* Module information */
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h
new file mode 100644
index 00000000000..91deadd5567
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.h
@@ -0,0 +1,47 @@
+/*
+ * ASoC PXA SSP port support
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA_SSP_H
+#define _PXA_SSP_H
+
+/* pxa DAI SSP IDs */
+#define PXA_DAI_SSP1			0
+#define PXA_DAI_SSP2			1
+#define PXA_DAI_SSP3			2
+#define PXA_DAI_SSP4			3
+
+/* SSP clock sources */
+#define PXA_SSP_CLK_PLL	0
+#define PXA_SSP_CLK_EXT	1
+#define PXA_SSP_CLK_NET	2
+#define PXA_SSP_CLK_AUDIO	3
+#define PXA_SSP_CLK_NET_PLL	4
+
+/* SSP audio dividers */
+#define PXA_SSP_AUDIO_DIV_ACDS		0
+#define PXA_SSP_AUDIO_DIV_SCDB		1
+#define PXA_SSP_DIV_SCR				2
+
+/* SSP ACDS audio dividers values */
+#define PXA_SSP_CLK_AUDIO_DIV_1		0
+#define PXA_SSP_CLK_AUDIO_DIV_2		1
+#define PXA_SSP_CLK_AUDIO_DIV_4		2
+#define PXA_SSP_CLK_AUDIO_DIV_8		3
+#define PXA_SSP_CLK_AUDIO_DIV_16	4
+#define PXA_SSP_CLK_AUDIO_DIV_32	5
+
+/* SSP divider bypass */
+#define PXA_SSP_CLK_SCDB_4		0
+#define PXA_SSP_CLK_SCDB_1		1
+#define PXA_SSP_CLK_SCDB_8		2
+
+#define PXA_SSP_PLL_OUT  0
+
+extern struct snd_soc_dai pxa_ssp_dai[4];
+
+#endif
-- 
cgit v1.2.3-70-g09d2


From 74e722015fe47c8f0e7ef7c0b4cf32d3e4ae11a0 Mon Sep 17 00:00:00 2001
From: Marek Vasut <marek.vasut@gmail.com>
Date: Mon, 3 Nov 2008 12:02:12 +0000
Subject: ASoC: Add Palm/PXA27x unified ASoC audio driver

this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for
palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test).
I sent it here some time ago, but now I got to fixing bugs in it. It should
be somehow mostly ok and ready for applying.

[Converted to use snd_soc_dapm_nc_pin() and bool Kconfig -- broonie]

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 arch/arm/mach-pxa/include/mach/palmasoc.h |  13 ++
 sound/soc/pxa/Kconfig                     |   9 +
 sound/soc/pxa/Makefile                    |   2 +
 sound/soc/pxa/palm27x.c                   | 269 ++++++++++++++++++++++++++++++
 4 files changed, 293 insertions(+)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c

(limited to 'sound/soc/pxa')

diff --git a/arch/arm/mach-pxa/include/mach/palmasoc.h b/arch/arm/mach-pxa/include/mach/palmasoc.h
new file mode 100644
index 00000000000..6c4b1f7de20
--- /dev/null
+++ b/arch/arm/mach-pxa/include/mach/palmasoc.h
@@ -0,0 +1,13 @@
+#ifndef _INCLUDE_PALMASOC_H_
+#define _INCLUDE_PALMASOC_H_
+struct palm27x_asoc_info {
+	int	jack_gpio;
+};
+
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);
+#else
+static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) {}
+#endif
+
+#endif
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 4235524238f..f82e1069947 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -79,6 +79,15 @@ config SND_PXA2XX_SOC_EM_X270
 	  Say Y if you want to add support for SoC audio on
 	  CompuLab EM-x270.
 
+config SND_PXA2XX_SOC_PALM27X
+	bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
+	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Palm T|X, T5 or LifeDrive handheld computer.
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 00258abb84a..1a3a36e75bf 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -16,6 +16,7 @@ snd-soc-tosa-objs := tosa.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -23,3 +24,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
new file mode 100644
index 00000000000..e364abc700d
--- /dev/null
+++ b/sound/soc/pxa/palm27x.c
@@ -0,0 +1,269 @@
+/*
+ * linux/sound/soc/pxa/palm27x.c
+ *
+ * SoC Audio driver for Palm T|X, T5 and LifeDrive
+ *
+ * based on tosa.c
+ *
+ * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/irq.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/audio.h>
+#include <mach/palmasoc.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int palm27x_jack_func = 1;
+static int palm27x_spk_func = 1;
+static int palm27x_ep_gpio = -1;
+
+static void palm27x_ext_control(struct snd_soc_codec *codec)
+{
+	if (!palm27x_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Speaker");
+	else
+		snd_soc_dapm_disable_pin(codec, "Speaker");
+
+	if (!palm27x_jack_func)
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+	snd_soc_dapm_sync(codec);
+}
+
+static int palm27x_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	palm27x_ext_control(codec);
+	return 0;
+}
+
+static struct snd_soc_ops palm27x_ops = {
+	.startup = palm27x_startup,
+};
+
+static irqreturn_t palm27x_interrupt(int irq, void *v)
+{
+	palm27x_spk_func = gpio_get_value(palm27x_ep_gpio);
+	palm27x_jack_func = !palm27x_spk_func;
+	return IRQ_HANDLED;
+}
+
+static int palm27x_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_jack_func;
+	return 0;
+}
+
+static int palm27x_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_jack_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+static int palm27x_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_spk_func;
+	return 0;
+}
+
+static int palm27x_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_spk_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+/* PalmTX machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* PalmTX audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to ROUT2, LOUT2 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+};
+
+static const char *jack_function[] = {"Headphone", "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum palm27x_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new palm27x_controls[] = {
+	SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack,
+		palm27x_set_jack),
+	SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk,
+		palm27x_set_spk),
+};
+
+static int palm27x_ac97_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+
+	/* add palm27x specific controls */
+	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&palm27x_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* add palm27x specific widgets */
+	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+				ARRAY_SIZE(palm27x_dapm_widgets));
+
+	/* set up palm27x specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+	.name = "AC97 HiFi",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.init = palm27x_ac97_init,
+	.ops = &palm27x_ops,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.ops = &palm27x_ops,
+},
+};
+
+static struct snd_soc_machine palm27x_asoc = {
+	.name = "Palm/PXA27x",
+	.dai_link = palm27x_dai,
+	.num_links = ARRAY_SIZE(palm27x_dai),
+};
+
+static struct snd_soc_device palm27x_snd_devdata = {
+	.machine = &palm27x_asoc,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *palm27x_snd_device;
+
+static int __init palm27x_asoc_init(void)
+{
+	int ret;
+
+	if (!(machine_is_palmtx() || machine_is_palmt5() ||
+		machine_is_palmld()))
+		return -ENODEV;
+
+	ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_input(palm27x_ep_gpio);
+	if (ret)
+		goto err_alloc;
+
+	if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt,
+			IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+			"Headphone jack", NULL))
+		goto err_alloc;
+
+	palm27x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!palm27x_snd_device) {
+		ret = -ENOMEM;
+		goto err_dev;
+	}
+
+	platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
+	palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
+	ret = platform_device_add(palm27x_snd_device);
+
+	if (ret != 0)
+		goto put_device;
+
+	return 0;
+
+put_device:
+	platform_device_put(palm27x_snd_device);
+err_dev:
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+err_alloc:
+	gpio_free(palm27x_ep_gpio);
+
+	return ret;
+}
+
+static void __exit palm27x_asoc_exit(void)
+{
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+	gpio_free(palm27x_ep_gpio);
+	platform_device_unregister(palm27x_snd_device);
+}
+
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+{
+	palm27x_ep_gpio = data->jack_gpio;
+}
+
+module_init(palm27x_asoc_init);
+module_exit(palm27x_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3-70-g09d2


From 875065491fba8eb13219f16c36e79a6fb4e15c68 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 18 Nov 2008 20:50:34 +0000
Subject: ASoC: Rename snd_soc_card to snd_soc_machine

One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.

Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 Documentation/sound/alsa/soc/machine.txt |  8 +--
 include/sound/soc.h                      |  8 +--
 sound/soc/atmel/playpaq_wm8510.c         |  4 +-
 sound/soc/atmel/sam9g20_wm8731.c         |  4 +-
 sound/soc/au1x/sample-ac97.c             |  4 +-
 sound/soc/blackfin/bf5xx-ad1980.c        |  6 +--
 sound/soc/blackfin/bf5xx-ad73311.c       |  6 +--
 sound/soc/blackfin/bf5xx-ssm2602.c       |  6 +--
 sound/soc/davinci/davinci-evm.c          |  4 +-
 sound/soc/davinci/davinci-i2s.c          |  8 +--
 sound/soc/davinci/davinci-sffsdr.c       |  4 +-
 sound/soc/fsl/fsl_dma.c                  |  2 +-
 sound/soc/fsl/mpc8610_hpcd.c             |  6 +--
 sound/soc/fsl/soc-of-simple.c            | 10 ++--
 sound/soc/omap/n810.c                    |  4 +-
 sound/soc/omap/omap2evm.c                |  4 +-
 sound/soc/omap/omap3beagle.c             |  4 +-
 sound/soc/omap/osk5912.c                 |  4 +-
 sound/soc/omap/overo.c                   |  4 +-
 sound/soc/pxa/corgi.c                    |  4 +-
 sound/soc/pxa/e800_wm9712.c              |  6 +--
 sound/soc/pxa/em-x270.c                  |  4 +-
 sound/soc/pxa/palm27x.c                  |  4 +-
 sound/soc/pxa/poodle.c                   |  4 +-
 sound/soc/pxa/spitz.c                    |  4 +-
 sound/soc/pxa/tosa.c                     |  6 +--
 sound/soc/s3c24xx/ln2440sbc_alc650.c     |  6 +--
 sound/soc/s3c24xx/neo1973_wm8753.c       |  6 +--
 sound/soc/s3c24xx/s3c24xx_uda134x.c      |  4 +-
 sound/soc/s3c24xx/smdk2443_wm9710.c      |  6 +--
 sound/soc/sh/sh7760-ac97.c               |  4 +-
 sound/soc/soc-core.c                     | 90 ++++++++++++++++----------------
 sound/soc/soc-dapm.c                     |  6 +--
 33 files changed, 127 insertions(+), 127 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index f370e7db86a..4a9f51e2905 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -9,7 +9,7 @@ the audio subsystem with the kernel as a platform device and is represented by
 the following struct:-
 
 /* SoC machine */
-struct snd_soc_machine {
+struct snd_soc_card {_
 	char *name;
 
 	int (*probe)(struct platform_device *pdev);
@@ -67,10 +67,10 @@ static struct snd_soc_dai_link corgi_dai = {
 	.ops = &corgi_ops,
 };
 
-struct snd_soc_machine then sets up the machine with it's DAIs. e.g.
+struct snd_soc_card then sets up the machine with it's DAIs. e.g.
 
 /* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
 	.dai_link = &corgi_dai,
 	.num_links = 1,
@@ -90,7 +90,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_machine_corgi,
+	.machine = &snd_soc_corgi,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 077dfe4e51f..3be17b3c650 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -482,8 +482,8 @@ struct snd_soc_dai_link  {
 	struct snd_pcm *pcm;
 };
 
-/* SoC machine */
-struct snd_soc_machine {
+/* SoC card */
+struct snd_soc_card {
 	char *name;
 
 	int (*probe)(struct platform_device *pdev);
@@ -497,7 +497,7 @@ struct snd_soc_machine {
 	int (*resume_post)(struct platform_device *pdev);
 
 	/* callbacks */
-	int (*set_bias_level)(struct snd_soc_machine *,
+	int (*set_bias_level)(struct snd_soc_card *,
 			      enum snd_soc_bias_level level);
 
 	/* CPU <--> Codec DAI links  */
@@ -508,7 +508,7 @@ struct snd_soc_machine {
 /* SoC Device - the audio subsystem */
 struct snd_soc_device {
 	struct device *dev;
-	struct snd_soc_machine *machine;
+	struct snd_soc_card *card;
 	struct snd_soc_platform *platform;
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index ea7935d2a66..d40b5a52a8d 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -361,7 +361,7 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = {
 
 
 
-static struct snd_soc_machine snd_soc_machine_playpaq = {
+static struct snd_soc_card snd_soc_playpaq = {
 	.name = "LRS_PlayPaq_WM8510",
 	.dai_link = &playpaq_wm8510_dai,
 	.num_links = 1,
@@ -377,7 +377,7 @@ static struct wm8510_setup_data playpaq_wm8510_setup = {
 
 
 static struct snd_soc_device playpaq_wm8510_snd_devdata = {
-	.machine = &snd_soc_machine_playpaq,
+	.card = &snd_soc_playpaq,
 	.platform = &at32_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8510,
 	.codec_data = &playpaq_wm8510_setup,
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 710addcc66b..fdc1d0206e0 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -242,7 +242,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
 	.ops = &at91sam9g20ek_ops,
 };
 
-static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = {
+static struct snd_soc_card snd_soc_at91sam9g20ek = {
 	.name = "WM8731",
 	.dai_link = &at91sam9g20ek_dai,
 	.num_links = 1,
@@ -254,7 +254,7 @@ static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
 };
 
 static struct snd_soc_device at91sam9g20ek_snd_devdata = {
-	.machine = &snd_soc_machine_at91sam9g20ek,
+	.card = &snd_soc_at91sam9g20ek,
 	.platform = &atmel_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &at91sam9g20ek_wm8731_setup,
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
index f75ae7f62c3..27683eb7905 100644
--- a/sound/soc/au1x/sample-ac97.c
+++ b/sound/soc/au1x/sample-ac97.c
@@ -42,14 +42,14 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
 	.ops		= NULL,
 };
 
-static struct snd_soc_machine au1xpsc_sample_ac97_machine = {
+static struct snd_soc_card au1xpsc_sample_ac97_machine = {
 	.name		= "Au1xxx PSC AC97 Audio",
 	.dai_link	= &au1xpsc_sample_ac97_dai,
 	.num_links	= 1,
 };
 
 static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
-	.machine	= &au1xpsc_sample_ac97_machine,
+	.card		= &au1xpsc_sample_ac97_machine,
 	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
 	.codec_dev	= &soc_codec_dev_ac97,
 };
diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c
index 124425d2232..36c569a43ce 100644
--- a/sound/soc/blackfin/bf5xx-ad1980.c
+++ b/sound/soc/blackfin/bf5xx-ad1980.c
@@ -43,7 +43,7 @@
 #include "bf5xx-ac97-pcm.h"
 #include "bf5xx-ac97.h"
 
-static struct snd_soc_machine bf5xx_board;
+static struct snd_soc_card bf5xx_board;
 
 static int bf5xx_board_startup(struct snd_pcm_substream *substream)
 {
@@ -67,14 +67,14 @@ static struct snd_soc_dai_link bf5xx_board_dai = {
 	.ops = &bf5xx_board_ops,
 };
 
-static struct snd_soc_machine bf5xx_board = {
+static struct snd_soc_card bf5xx_board = {
 	.name = "bf5xx-board",
 	.dai_link = &bf5xx_board_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_board_snd_devdata = {
-	.machine = &bf5xx_board,
+	.card = &bf5xx_board,
 	.platform = &bf5xx_ac97_soc_platform,
 	.codec_dev = &soc_codec_dev_ad1980,
 };
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 47da49b9aea..57da1479937 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -65,7 +65,7 @@
 
 #define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
 
-static struct snd_soc_machine bf5xx_ad73311;
+static struct snd_soc_card bf5xx_ad73311;
 
 static int snd_ad73311_startup(void)
 {
@@ -190,7 +190,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = {
 	.ops = &bf5xx_ad73311_ops,
 };
 
-static struct snd_soc_machine bf5xx_ad73311 = {
+static struct snd_soc_card bf5xx_ad73311 = {
 	.name = "bf5xx_ad73311",
 	.probe = bf5xx_probe,
 	.dai_link = &bf5xx_ad73311_dai,
@@ -198,7 +198,7 @@ static struct snd_soc_machine bf5xx_ad73311 = {
 };
 
 static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
-	.machine = &bf5xx_ad73311,
+	.card = &bf5xx_ad73311,
 	.platform = &bf5xx_i2s_soc_platform,
 	.codec_dev = &soc_codec_dev_ad73311,
 };
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index 744a90e765d..0078dfcd95b 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -44,7 +44,7 @@
 #include "bf5xx-i2s-pcm.h"
 #include "bf5xx-i2s.h"
 
-static struct snd_soc_machine bf5xx_ssm2602;
+static struct snd_soc_card bf5xx_ssm2602;
 
 static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream)
 {
@@ -135,14 +135,14 @@ static struct ssm2602_setup_data bf5xx_ssm2602_setup = {
 	.i2c_address = 0x1b,
 };
 
-static struct snd_soc_machine bf5xx_ssm2602 = {
+static struct snd_soc_card bf5xx_ssm2602 = {
 	.name = "bf5xx_ssm2602",
 	.dai_link = &bf5xx_ssm2602_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
-	.machine = &bf5xx_ssm2602,
+	.card = &bf5xx_ssm2602,
 	.platform = &bf5xx_i2s_soc_platform,
 	.codec_dev = &soc_codec_dev_ssm2602,
 	.codec_data = &bf5xx_ssm2602_setup,
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9e6062cd6b5..2ce34d44b15 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -128,7 +128,7 @@ static struct snd_soc_dai_link evm_dai = {
 };
 
 /* davinci-evm audio machine driver */
-static struct snd_soc_machine snd_soc_machine_evm = {
+static struct snd_soc_card snd_soc_card_evm = {
 	.name = "DaVinci EVM",
 	.dai_link = &evm_dai,
 	.num_links = 1,
@@ -142,7 +142,7 @@ static struct aic3x_setup_data evm_aic3x_setup = {
 
 /* evm audio subsystem */
 static struct snd_soc_device evm_snd_devdata = {
-	.machine = &snd_soc_machine_evm,
+	.card = &snd_soc_card_evm,
 	.platform = &davinci_soc_platform,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &evm_aic3x_setup,
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 11c20d0b7bc..95df51e803b 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -375,8 +375,8 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *ioarea;
 	struct evm_snd_platform_data *pdata;
@@ -437,8 +437,8 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
 	struct resource *mem;
 
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 69a8a769f4d..fa38f9cd350 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -73,7 +73,7 @@ static struct snd_soc_dai_link sffsdr_dai = {
 };
 
 /* davinci-sffsdr audio machine driver */
-static struct snd_soc_machine snd_soc_machine_sffsdr = {
+static struct snd_soc_card snd_soc_sffsdr = {
 	.name = "DaVinci SFFSDR",
 	.dai_link = &sffsdr_dai,
 	.num_links = 1,
@@ -89,7 +89,7 @@ static struct pcm3008_setup_data sffsdr_pcm3008_setup = {
 
 /* sffsdr audio subsystem */
 static struct snd_soc_device sffsdr_snd_devdata = {
-	.machine = &snd_soc_machine_sffsdr,
+	.card = &snd_soc_sffsdr,
 	.platform = &davinci_soc_platform,
 	.codec_dev = &soc_codec_dev_pcm3008,
 	.codec_data = &sffsdr_pcm3008_setup,
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index d2d3da9729f..bf92331b476 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -284,7 +284,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
  * fsl_dma_new: initialize this PCM driver.
  *
  * This function is called when the codec driver calls snd_soc_new_pcms(),
- * once for each .dai_link in the machine driver's snd_soc_machine
+ * once for each .dai_link in the machine driver's snd_soc_card
  * structure.
  */
 static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 94f89debde1..1cf4d6eeb53 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -29,7 +29,7 @@
 struct mpc8610_hpcd_data {
 	struct snd_soc_device sound_devdata;
 	struct snd_soc_dai_link dai;
-	struct snd_soc_machine machine;
+	struct snd_soc_card machine;
 	unsigned int dai_format;
 	unsigned int codec_clk_direction;
 	unsigned int cpu_clk_direction;
@@ -185,7 +185,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 /**
  * mpc8610_hpcd_machine: ASoC machine data
  */
-static struct snd_soc_machine mpc8610_hpcd_machine = {
+static struct snd_soc_card mpc8610_hpcd_machine = {
 	.probe = mpc8610_hpcd_machine_probe,
 	.remove = mpc8610_hpcd_machine_remove,
 	.name = "MPC8610 HPCD",
@@ -465,7 +465,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 		goto error;
 	}
 
-	machine_data->sound_devdata.machine = &mpc8610_hpcd_machine;
+	machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
 	machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
 	machine_data->sound_devdata.platform = &fsl_soc_platform;
 
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c
index 0382fdac51c..53be6491320 100644
--- a/sound/soc/fsl/soc-of-simple.c
+++ b/sound/soc/fsl/soc-of-simple.c
@@ -31,7 +31,7 @@ struct of_snd_soc_device {
 	int id;
 	struct list_head list;
 	struct snd_soc_device device;
-	struct snd_soc_machine machine;
+	struct snd_soc_card card;
 	struct snd_soc_dai_link dai_link;
 	struct platform_device *pdev;
 	struct device_node *platform_node;
@@ -58,9 +58,9 @@ of_snd_soc_get_device(struct device_node *codec_node)
 	/* Initialize the structure and add it to the global list */
 	of_soc->codec_node = codec_node;
 	of_soc->id = of_snd_soc_next_index++;
-	of_soc->machine.dai_link = &of_soc->dai_link;
-	of_soc->machine.num_links = 1;
-	of_soc->device.machine = &of_soc->machine;
+	of_soc->card.dai_link = &of_soc->dai_link;
+	of_soc->card.num_links = 1;
+	of_soc->device.card = &of_soc->card;
 	of_soc->dai_link.ops = &of_snd_soc_ops;
 	list_add(&of_soc->list, &of_snd_soc_device_list);
 
@@ -159,7 +159,7 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform,
 	of_soc->platform_node = node;
 	of_soc->dai_link.cpu_dai = cpu_dai;
 	of_soc->device.platform = platform;
-	of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+	of_soc->card.name = of_soc->dai_link.cpu_dai->name;
 
 	/* Now try to register the SoC device */
 	of_snd_soc_register_device(of_soc);
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index fae3ad36e0b..d216b4f9e14 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -282,7 +282,7 @@ static struct snd_soc_dai_link n810_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_n810 = {
+static struct snd_soc_card snd_soc_n810 = {
 	.name = "N810",
 	.dai_link = &n810_dai,
 	.num_links = 1,
@@ -298,7 +298,7 @@ static struct aic3x_setup_data n810_aic33_setup = {
 
 /* Audio subsystem */
 static struct snd_soc_device n810_snd_devdata = {
-	.machine = &snd_soc_machine_n810,
+	.card = &snd_soc_n810,
 	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &n810_aic33_setup,
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index c37621357d0..5bea31157a9 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -90,7 +90,7 @@ static struct snd_soc_dai_link omap2evm_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_omap2evm = {
+static struct snd_soc_card snd_soc_omap2evm = {
 	.name = "omap2evm",
 	.dai_link = &omap2evm_dai,
 	.num_links = 1,
@@ -98,7 +98,7 @@ static struct snd_soc_machine snd_soc_machine_omap2evm = {
 
 /* Audio subsystem */
 static struct snd_soc_device omap2evm_snd_devdata = {
-	.machine = &snd_soc_machine_omap2evm,
+	.card = &snd_soc_omap2evm,
 	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index ec84a9bbc56..3ed25464627 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -88,7 +88,7 @@ static struct snd_soc_dai_link omap3beagle_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_omap3beagle = {
+static struct snd_soc_card snd_soc_omap3beagle = {
 	.name = "omap3beagle",
 	.dai_link = &omap3beagle_dai,
 	.num_links = 1,
@@ -96,7 +96,7 @@ static struct snd_soc_machine snd_soc_machine_omap3beagle = {
 
 /* Audio subsystem */
 static struct snd_soc_device omap3beagle_snd_devdata = {
-	.machine = &snd_soc_machine_omap3beagle,
+	.card = &snd_soc_omap3beagle,
 	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 0fe73379689..7a8f14d0c77 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -143,7 +143,7 @@ static struct snd_soc_dai_link osk_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_osk = {
+static struct snd_soc_card snd_soc_card_osk = {
 	.name = "OSK5912",
 	.dai_link = &osk_dai,
 	.num_links = 1,
@@ -151,7 +151,7 @@ static struct snd_soc_machine snd_soc_machine_osk = {
 
 /* Audio subsystem */
 static struct snd_soc_device osk_snd_devdata = {
-	.machine = &snd_soc_machine_osk,
+	.card = &snd_soc_card_osk,
 	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_tlv320aic23,
 };
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index c26d1de7da5..eea0c372bb3 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -88,7 +88,7 @@ static struct snd_soc_dai_link overo_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_overo = {
+static struct snd_soc_card snd_soc_card_overo = {
 	.name = "overo",
 	.dai_link = &overo_dai,
 	.num_links = 1,
@@ -96,7 +96,7 @@ static struct snd_soc_machine snd_soc_machine_overo = {
 
 /* Audio subsystem */
 static struct snd_soc_device overo_snd_devdata = {
-	.machine = &snd_soc_machine_overo,
+	.card = &snd_soc_card_overo,
 	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 2718eaf7895..647f056a3cb 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -314,7 +314,7 @@ static struct snd_soc_dai_link corgi_dai = {
 };
 
 /* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
 	.dai_link = &corgi_dai,
 	.num_links = 1,
@@ -328,7 +328,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_machine_corgi,
+	.card = &snd_soc_corgi,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 6781c5be242..60c64861512 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -29,7 +29,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine e800;
+static struct snd_soc_card e800;
 
 static struct snd_soc_dai_link e800_dai[] = {
 {
@@ -40,14 +40,14 @@ static struct snd_soc_dai_link e800_dai[] = {
 },
 };
 
-static struct snd_soc_machine e800 = {
+static struct snd_soc_card e800 = {
 	.name = "Toshiba e800",
 	.dai_link = e800_dai,
 	.num_links = ARRAY_SIZE(e800_dai),
 };
 
 static struct snd_soc_device e800_snd_devdata = {
-	.machine = &e800,
+	.card = &e800,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index e6ff6929ab4..4a61925c310 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -53,14 +53,14 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 	},
 };
 
-static struct snd_soc_machine em_x270 = {
+static struct snd_soc_card em_x270 = {
 	.name = "EM-X270",
 	.dai_link = em_x270_dai,
 	.num_links = ARRAY_SIZE(em_x270_dai),
 };
 
 static struct snd_soc_device em_x270_snd_devdata = {
-	.machine = &em_x270,
+	.card = &em_x270,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index e364abc700d..3bb8879ac8a 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -189,14 +189,14 @@ static struct snd_soc_dai_link palm27x_dai[] = {
 },
 };
 
-static struct snd_soc_machine palm27x_asoc = {
+static struct snd_soc_card palm27x_asoc = {
 	.name = "Palm/PXA27x",
 	.dai_link = palm27x_dai,
 	.num_links = ARRAY_SIZE(palm27x_dai),
 };
 
 static struct snd_soc_device palm27x_snd_devdata = {
-	.machine = &palm27x_asoc,
+	.card = &palm27x_asoc,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 4d9930c5278..03b510ab282 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
 };
 
 /* poodle audio machine driver */
-static struct snd_soc_machine snd_soc_machine_poodle = {
+static struct snd_soc_card snd_soc_poodle = {
 	.name = "Poodle",
 	.dai_link = &poodle_dai,
 	.num_links = 1,
@@ -290,7 +290,7 @@ static struct wm8731_setup_data poodle_wm8731_setup = {
 
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
-	.machine = &snd_soc_machine_poodle,
+	.card = &snd_soc_poodle,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &poodle_wm8731_setup,
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d307b6757e9..579d93368f1 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -319,7 +319,7 @@ static struct snd_soc_dai_link spitz_dai = {
 };
 
 /* spitz audio machine driver */
-static struct snd_soc_machine snd_soc_machine_spitz = {
+static struct snd_soc_card snd_soc_spitz = {
 	.name = "Spitz",
 	.dai_link = &spitz_dai,
 	.num_links = 1,
@@ -333,7 +333,7 @@ static struct wm8750_setup_data spitz_wm8750_setup = {
 
 /* spitz audio subsystem */
 static struct snd_soc_device spitz_snd_devdata = {
-	.machine = &snd_soc_machine_spitz,
+	.card = &snd_soc_spitz,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8750,
 	.codec_data = &spitz_wm8750_setup,
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index afefe41b8c4..9d9be5a14d1 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -38,7 +38,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine tosa;
+static struct snd_soc_card tosa;
 
 #define TOSA_HP        0
 #define TOSA_MIC_INT   1
@@ -230,14 +230,14 @@ static struct snd_soc_dai_link tosa_dai[] = {
 },
 };
 
-static struct snd_soc_machine tosa = {
+static struct snd_soc_card tosa = {
 	.name = "Tosa",
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
 };
 
 static struct snd_soc_device tosa_snd_devdata = {
-	.machine = &tosa,
+	.card = &tosa,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
index 4eab2c19c45..a70cbc0fa07 100644
--- a/sound/soc/s3c24xx/ln2440sbc_alc650.c
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -27,7 +27,7 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
 
-static struct snd_soc_machine ln2440sbc;
+static struct snd_soc_card ln2440sbc;
 
 static struct snd_soc_dai_link ln2440sbc_dai[] = {
 {
@@ -38,14 +38,14 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = {
 },
 };
 
-static struct snd_soc_machine ln2440sbc = {
+static struct snd_soc_card ln2440sbc = {
 	.name = "LN2440SBC",
 	.dai_link = ln2440sbc_dai,
 	.num_links = ARRAY_SIZE(ln2440sbc_dai),
 };
 
 static struct snd_soc_device ln2440sbc_snd_ac97_devdata = {
-	.machine = &ln2440sbc,
+	.card = &ln2440sbc,
 	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_ac97,
 };
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 87ddfefcc2f..528fc3f1b45 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -59,7 +59,7 @@
 #define NEO_CAPTURE_HEADSET		7
 #define NEO_CAPTURE_BLUETOOTH		8
 
-static struct snd_soc_machine neo1973;
+static struct snd_soc_card neo1973;
 static struct i2c_client *i2c;
 
 static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
@@ -579,7 +579,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 },
 };
 
-static struct snd_soc_machine neo1973 = {
+static struct snd_soc_card neo1973 = {
 	.name = "neo1973",
 	.dai_link = neo1973_dai,
 	.num_links = ARRAY_SIZE(neo1973_dai),
@@ -591,7 +591,7 @@ static struct wm8753_setup_data neo1973_wm8753_setup = {
 };
 
 static struct snd_soc_device neo1973_snd_devdata = {
-	.machine = &neo1973,
+	.card = &neo1973,
 	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8753,
 	.codec_data = &neo1973_wm8753_setup,
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index 92d90f2f690..23325fca1f6 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -232,7 +232,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
 	.ops = &s3c24xx_uda134x_ops,
 };
 
-static struct snd_soc_machine snd_soc_machine_s3c24xx_uda134x = {
+static struct snd_soc_card snd_soc_s3c24xx_uda134x = {
 	.name = "S3C24XX_UDA134X",
 	.dai_link = &s3c24xx_uda134x_dai_link,
 	.num_links = 1,
@@ -270,7 +270,7 @@ static struct uda134x_platform_data s3c24xx_uda134x = {
 };
 
 static struct snd_soc_device s3c24xx_uda134x_snd_devdata = {
-	.machine = &snd_soc_machine_s3c24xx_uda134x,
+	.card = &snd_soc_s3c24xx_uda134x,
 	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_uda134x,
 	.codec_data = &s3c24xx_uda134x,
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index 8515d6ff03f..3d2e6a0417e 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -23,7 +23,7 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
 
-static struct snd_soc_machine smdk2443;
+static struct snd_soc_card smdk2443;
 
 static struct snd_soc_dai_link smdk2443_dai[] = {
 {
@@ -34,14 +34,14 @@ static struct snd_soc_dai_link smdk2443_dai[] = {
 },
 };
 
-static struct snd_soc_machine smdk2443 = {
+static struct snd_soc_card smdk2443 = {
 	.name = "SMDK2443",
 	.dai_link = smdk2443_dai,
 	.num_links = ARRAY_SIZE(smdk2443_dai),
 };
 
 static struct snd_soc_device smdk2443_snd_ac97_devdata = {
-	.machine = &smdk2443,
+	.card = &smdk2443,
 	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_ac97,
 };
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 92bfaf4774a..8b44f9c8a9f 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -38,14 +38,14 @@ static struct snd_soc_dai_link sh7760_ac97_dai = {
 	.ops = NULL,
 };
 
-static struct snd_soc_machine sh7760_ac97_soc_machine  = {
+static struct snd_soc_card sh7760_ac97_soc_machine  = {
 	.name = "SH7760 AC97",
 	.dai_link = &sh7760_ac97_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device sh7760_ac97_snd_devdata = {
-	.machine = &sh7760_ac97_soc_machine,
+	.card = &sh7760_ac97_soc_machine,
 	.platform = &sh7760_soc_platform,
 	.codec_dev = &soc_codec_dev_ac97,
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 9feaa7b6dc3..c5cb9516fea 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -620,7 +620,7 @@ static struct snd_pcm_ops soc_pcm_ops = {
 static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = socdev->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -644,14 +644,14 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	}
 
 	/* suspend all pcms */
-	for (i = 0; i < machine->num_links; i++)
-		snd_pcm_suspend_all(machine->dai_link[i].pcm);
+	for (i = 0; i < card->num_links; i++)
+		snd_pcm_suspend_all(card->dai_link[i].pcm);
 
-	if (machine->suspend_pre)
-		machine->suspend_pre(pdev, state);
+	if (card->suspend_pre)
+		card->suspend_pre(pdev, state);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai  *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
 			cpu_dai->suspend(pdev, cpu_dai);
 		if (platform->suspend)
@@ -676,14 +676,14 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	if (codec_dev->suspend)
 		codec_dev->suspend(pdev, state);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
 			cpu_dai->suspend(pdev, cpu_dai);
 	}
 
-	if (machine->suspend_post)
-		machine->suspend_post(pdev, state);
+	if (card->suspend_post)
+		card->suspend_post(pdev, state);
 
 	return 0;
 }
@@ -696,7 +696,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	struct snd_soc_device *socdev = container_of(work,
 						     struct snd_soc_device,
 						     deferred_resume_work);
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = socdev->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -709,11 +709,11 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	dev_info(socdev->dev, "starting resume work\n");
 
-	if (machine->resume_pre)
-		machine->resume_pre(pdev);
+	if (card->resume_pre)
+		card->resume_pre(pdev);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
 			cpu_dai->resume(pdev, cpu_dai);
 	}
@@ -739,16 +739,16 @@ static void soc_resume_deferred(struct work_struct *work)
 			dai->dai_ops.digital_mute(dai, 0);
 	}
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
 			cpu_dai->resume(pdev, cpu_dai);
 		if (platform->resume)
 			platform->resume(pdev, cpu_dai);
 	}
 
-	if (machine->resume_post)
-		machine->resume_post(pdev);
+	if (card->resume_post)
+		card->resume_post(pdev);
 
 	dev_info(socdev->dev, "resume work completed\n");
 
@@ -779,18 +779,18 @@ static int soc_probe(struct platform_device *pdev)
 {
 	int ret = 0, i;
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = socdev->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
-	if (machine->probe) {
-		ret = machine->probe(pdev);
+	if (card->probe) {
+		ret = card->probe(pdev);
 		if (ret < 0)
 			return ret;
 	}
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->probe) {
 			ret = cpu_dai->probe(pdev, cpu_dai);
 			if (ret < 0)
@@ -825,13 +825,13 @@ platform_err:
 
 cpu_dai_err:
 	for (i--; i >= 0; i--) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
 			cpu_dai->remove(pdev, cpu_dai);
 	}
 
-	if (machine->remove)
-		machine->remove(pdev);
+	if (card->remove)
+		card->remove(pdev);
 
 	return ret;
 }
@@ -841,7 +841,7 @@ static int soc_remove(struct platform_device *pdev)
 {
 	int i;
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = socdev->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
@@ -853,14 +853,14 @@ static int soc_remove(struct platform_device *pdev)
 	if (codec_dev->remove)
 		codec_dev->remove(pdev);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
 			cpu_dai->remove(pdev, cpu_dai);
 	}
 
-	if (machine->remove)
-		machine->remove(pdev);
+	if (card->remove)
+		card->remove(pdev);
 
 	return 0;
 }
@@ -1212,7 +1212,7 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0, i;
 
 	mutex_lock(&codec->mutex);
@@ -1231,11 +1231,11 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
 
 	/* create the pcms */
-	for (i = 0; i < machine->num_links; i++) {
-		ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
+	for (i = 0; i < card->num_links; i++) {
+		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't create pcm %s\n",
-				machine->dai_link[i].stream_name);
+				card->dai_link[i].stream_name);
 			mutex_unlock(&codec->mutex);
 			return ret;
 		}
@@ -1258,26 +1258,26 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
 int snd_soc_register_card(struct snd_soc_device *socdev)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0, i, ac97 = 0, err = 0;
 
-	for (i = 0; i < machine->num_links; i++) {
-		if (socdev->machine->dai_link[i].init) {
-			err = socdev->machine->dai_link[i].init(codec);
+	for (i = 0; i < card->num_links; i++) {
+		if (card->dai_link[i].init) {
+			err = card->dai_link[i].init(codec);
 			if (err < 0) {
 				printk(KERN_ERR "asoc: failed to init %s\n",
-					socdev->machine->dai_link[i].stream_name);
+					card->dai_link[i].stream_name);
 				continue;
 			}
 		}
-		if (socdev->machine->dai_link[i].codec_dai->type ==
+		if (card->dai_link[i].codec_dai->type ==
 			SND_SOC_DAI_AC97_BUS)
 			ac97 = 1;
 	}
 	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
-		 "%s", machine->name);
+		 "%s",  card->name);
 	snprintf(codec->card->longname, sizeof(codec->card->longname),
-		 "%s (%s)", machine->name, codec->name);
+		 "%s (%s)", card->name, codec->name);
 
 	ret = snd_card_register(codec->card);
 	if (ret < 0) {
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0fecbb44726..61d7d85aa57 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1402,11 +1402,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 				enum snd_soc_bias_level level)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0;
 
-	if (machine->set_bias_level)
-		ret = machine->set_bias_level(machine, level);
+	if (card->set_bias_level)
+		ret = card->set_bias_level(card, level);
 	if (ret == 0 && codec->set_bias_level)
 		ret = codec->set_bias_level(codec, level);
 
-- 
cgit v1.2.3-70-g09d2


From dee89c4d94433520e4e3977ae203d4cfbfe385fb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 18 Nov 2008 22:11:38 +0000
Subject: ASoC: Merge snd_soc_ops into snd_soc_dai_ops

Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.

This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h            | 20 +++++++++--
 sound/soc/atmel/atmel_ssc_dai.c    | 21 +++++------
 sound/soc/au1x/psc-ac97.c          |  5 +--
 sound/soc/au1x/psc-i2s.c           |  8 ++---
 sound/soc/blackfin/bf5xx-i2s.c     | 12 ++++---
 sound/soc/codecs/ac97.c            |  3 +-
 sound/soc/codecs/ak4535.c          |  5 ++-
 sound/soc/codecs/cs4270.c          | 11 +++---
 sound/soc/codecs/ssm2602.c         | 14 ++++----
 sound/soc/codecs/tlv320aic23.c     | 19 +++++-----
 sound/soc/codecs/tlv320aic26.c     |  5 ++-
 sound/soc/codecs/tlv320aic3x.c     |  5 ++-
 sound/soc/codecs/twl4030.c         |  5 ++-
 sound/soc/codecs/uda134x.c         | 12 +++----
 sound/soc/codecs/uda1380.c         | 15 ++++----
 sound/soc/codecs/wm8510.c          |  5 ++-
 sound/soc/codecs/wm8580.c          | 12 +++----
 sound/soc/codecs/wm8728.c          |  5 ++-
 sound/soc/codecs/wm8731.c          | 11 +++---
 sound/soc/codecs/wm8750.c          |  5 ++-
 sound/soc/codecs/wm8753.c          | 25 ++++++-------
 sound/soc/codecs/wm8900.c          |  5 ++-
 sound/soc/codecs/wm8903.c          | 11 +++---
 sound/soc/codecs/wm8971.c          |  5 ++-
 sound/soc/codecs/wm8990.c          |  6 ++--
 sound/soc/codecs/wm9712.c          |  6 ++--
 sound/soc/codecs/wm9713.c          | 21 +++++------
 sound/soc/davinci/davinci-i2s.c    | 12 ++++---
 sound/soc/davinci/davinci-sffsdr.c |  3 +-
 sound/soc/fsl/fsl_ssi.c            | 14 ++++----
 sound/soc/fsl/mpc5200_psc_i2s.c    | 17 +++++----
 sound/soc/omap/omap-mcbsp.c        | 14 ++++----
 sound/soc/omap/omap2evm.c          |  3 +-
 sound/soc/pxa/pxa-ssp.c            | 20 +++++------
 sound/soc/pxa/pxa2xx-ac97.c        |  9 +++--
 sound/soc/pxa/pxa2xx-i2s.c         | 15 ++++----
 sound/soc/s3c24xx/s3c2412-i2s.c    |  8 ++---
 sound/soc/s3c24xx/s3c2443-ac97.c   |  8 +++--
 sound/soc/s3c24xx/s3c24xx-i2s.c    |  9 ++---
 sound/soc/sh/hac.c                 |  3 +-
 sound/soc/sh/ssi.c                 | 16 ++++-----
 sound/soc/soc-core.c               | 74 +++++++++++++++++++-------------------
 42 files changed, 265 insertions(+), 237 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 08b8f7025c6..f51cb55902f 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -156,6 +156,23 @@ struct snd_soc_dai_ops {
 	 * Called by soc-core to minimise any pops.
 	 */
 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+
+	/*
+	 * ALSA PCM audio operations - all optional.
+	 * Called by soc-core during audio PCM operations.
+	 */
+	int (*startup)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	void (*shutdown)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*hw_params)(struct snd_pcm_substream *,
+		struct snd_pcm_hw_params *, struct snd_soc_dai *);
+	int (*hw_free)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*prepare)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*trigger)(struct snd_pcm_substream *, int,
+		struct snd_soc_dai *);
 };
 
 /*
@@ -180,8 +197,7 @@ struct snd_soc_dai {
 		struct snd_soc_dai *dai);
 
 	/* ops */
-	struct snd_soc_ops ops;
-	struct snd_soc_dai_ops dai_ops;
+	struct snd_soc_dai_ops ops;
 
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index d290b789491..916f73b9a18 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -202,7 +202,8 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id)
 /*
  * Startup.  Only that one substream allowed in each direction.
  */
-static int atmel_ssc_startup(struct snd_pcm_substream *substream)
+static int atmel_ssc_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
@@ -231,7 +232,8 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream)
  * Shutdown.  Clear DMA parameters and shutdown the SSC if there
  * are no other substreams open.
  */
-static void atmel_ssc_shutdown(struct snd_pcm_substream *substream)
+static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
@@ -332,7 +334,8 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
  * Configure the SSC.
  */
 static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	int id = rtd->dai->cpu_dai->id;
@@ -600,7 +603,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 }
 
 
-static int atmel_ssc_prepare(struct snd_pcm_substream *substream)
+static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
@@ -715,8 +719,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.startup = atmel_ssc_startup,
 			.shutdown = atmel_ssc_shutdown,
 			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,},
-		.dai_ops = {
+			.hw_params = atmel_ssc_hw_params,
 			.set_fmt = atmel_ssc_set_dai_fmt,
 			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
 		.private_data = &ssc_info[0],
@@ -741,8 +744,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.startup = atmel_ssc_startup,
 			.shutdown = atmel_ssc_shutdown,
 			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,},
-		.dai_ops = {
+			.hw_params = atmel_ssc_hw_params,
 			.set_fmt = atmel_ssc_set_dai_fmt,
 			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
 		.private_data = &ssc_info[1],
@@ -766,8 +768,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.startup = atmel_ssc_startup,
 			.shutdown = atmel_ssc_shutdown,
 			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,},
-		.dai_ops = {
+			.hw_params = atmel_ssc_hw_params,
 			.set_fmt = atmel_ssc_set_dai_fmt,
 			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
 		.private_data = &ssc_info[2],
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 57facbad682..ad60a6042ca 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -160,7 +160,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
-				  struct snd_pcm_hw_params *params)
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -210,7 +211,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
-				int cmd)
+				int cmd, struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 9384702c7eb..05a5acbb16a 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -116,7 +116,8 @@ out:
 }
 
 static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 
@@ -240,7 +241,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
 	return 0;
 }
 
-static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 	int ret, stype = SUBSTREAM_TYPE(substream);
@@ -389,8 +391,6 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 	.ops = {
 		.trigger	= au1xpsc_i2s_trigger,
 		.hw_params	= au1xpsc_i2s_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt	= au1xpsc_i2s_set_fmt,
 	},
 };
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index e020c160ee4..4e675b55b18 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -132,7 +132,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	return ret;
 }
 
-static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
+static int bf5xx_i2s_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 
@@ -142,7 +143,8 @@ static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
 }
 
 static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -193,7 +195,8 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	bf5xx_i2s.counter--;
@@ -307,8 +310,7 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 	.ops = {
 		.startup   = bf5xx_i2s_startup,
 		.shutdown  = bf5xx_i2s_shutdown,
-		.hw_params = bf5xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = bf5xx_i2s_hw_params,
 		.set_fmt = bf5xx_i2s_set_dai_fmt,
 	},
 };
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index bd1ebdc6c86..8a93aff359d 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -24,7 +24,8 @@
 
 #define AC97_VERSION "0.6"
 
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 2a89b5888e1..c742290e553 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -339,7 +339,8 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 }
 
 static int ak4535_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -451,8 +452,6 @@ struct snd_soc_dai ak4535_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
 		.hw_params = ak4535_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt = ak4535_set_dai_fmt,
 		.digital_mute = ak4535_mute,
 		.set_sysclk = ak4535_set_dai_sysclk,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0ff476d7057..7507d468b20 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -360,13 +360,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 /*
  * Program the CS4270 with the given hardware parameters.
  *
- * The .dai_ops functions are used to provide board-specific data, like
+ * The .ops functions are used to provide board-specific data, like
  * input frequencies, to this driver.  This function takes that information,
  * combines it with the hardware parameters provided, and programs the
  * hardware accordingly.
  */
 static int cs4270_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -710,10 +711,10 @@ static int cs4270_probe(struct platform_device *pdev)
 	if (codec->control_data) {
 		/* Initialize codec ops */
 		cs4270_dai.ops.hw_params = cs4270_hw_params;
-		cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk;
-		cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt;
+		cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
+		cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
 #ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-		cs4270_dai.dai_ops.digital_mute = cs4270_mute;
+		cs4270_dai.ops.digital_mute = cs4270_mute;
 #endif
 	} else
 		printk(KERN_INFO "cs4270: no I2C device found, "
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 56dc1c9c7c5..0c5884ea1b0 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -285,7 +285,8 @@ static inline int get_coeff(int mclk, int rate)
 }
 
 static int ssm2602_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	u16 srate;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -330,7 +331,8 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int ssm2602_startup(struct snd_pcm_substream *substream)
+static int ssm2602_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -366,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
+static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -377,7 +380,8 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void ssm2602_shutdown(struct snd_pcm_substream *substream)
+static void ssm2602_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -536,8 +540,6 @@ struct snd_soc_dai ssm2602_dai = {
 		.prepare = ssm2602_pcm_prepare,
 		.hw_params = ssm2602_hw_params,
 		.shutdown = ssm2602_shutdown,
-	},
-	.dai_ops = {
 		.digital_mute = ssm2602_mute,
 		.set_sysclk = ssm2602_set_dai_sysclk,
 		.set_fmt = ssm2602_set_dai_fmt,
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index c903e4f48dc..a4e13d0688c 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -418,7 +418,8 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -465,7 +466,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -477,7 +479,8 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -613,12 +616,10 @@ struct snd_soc_dai tlv320aic23_dai = {
 		.prepare = tlv320aic23_pcm_prepare,
 		.hw_params = tlv320aic23_hw_params,
 		.shutdown = tlv320aic23_shutdown,
-		},
-	.dai_ops = {
-		    .digital_mute = tlv320aic23_mute,
-		    .set_fmt = tlv320aic23_set_dai_fmt,
-		    .set_sysclk = tlv320aic23_set_dai_sysclk,
-		    }
+		.digital_mute = tlv320aic23_mute,
+		.set_fmt = tlv320aic23_set_dai_fmt,
+		.set_sysclk = tlv320aic23_set_dai_sysclk,
+	}
 };
 EXPORT_SYMBOL_GPL(tlv320aic23_dai);
 
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index bed8a9e63dd..6b7ddfc9257 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -125,7 +125,8 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
  * Digital Audio Interface Operations
  */
 static int aic26_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params)
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -287,8 +288,6 @@ struct snd_soc_dai aic26_dai = {
 	},
 	.ops = {
 		.hw_params = aic26_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = aic26_mute,
 		.set_sysclk = aic26_set_sysclk,
 		.set_fmt = aic26_set_fmt,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index cff276ee261..b76bcc3c411 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -694,7 +694,8 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int aic3x_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params)
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1009,8 +1010,6 @@ struct snd_soc_dai aic3x_dai = {
 		.formats = AIC3X_FORMATS,},
 	.ops = {
 		.hw_params = aic3x_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = aic3x_mute,
 		.set_sysclk = aic3x_set_dai_sysclk,
 		.set_fmt = aic3x_set_dai_fmt,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index c778eb446a5..33489515b92 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -343,7 +343,8 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 }
 
 static int twl4030_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params)
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -523,8 +524,6 @@ struct snd_soc_dai twl4030_dai = {
 		.formats = TWL4030_FORMATS,},
 	.ops = {
 		.hw_params = twl4030_hw_params,
-	},
-	.dai_ops = {
 		.set_sysclk = twl4030_set_dai_sysclk,
 		.set_fmt = twl4030_set_dai_fmt,
 	}
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 69ef521a2ed..91f333cdc7c 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -168,7 +168,8 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute)
 	return 0;
 }
 
-static int uda134x_startup(struct snd_pcm_substream *substream)
+static int uda134x_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -200,7 +201,8 @@ static int uda134x_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void uda134x_shutdown(struct snd_pcm_substream *substream)
+static void uda134x_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -214,7 +216,8 @@ static void uda134x_shutdown(struct snd_pcm_substream *substream)
 }
 
 static int uda134x_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -484,9 +487,6 @@ struct snd_soc_dai uda134x_dai = {
 		.startup = uda134x_startup,
 		.shutdown = uda134x_shutdown,
 		.hw_params = uda134x_hw_params,
-	},
-	/* DAI operations */
-	.dai_ops = {
 		.digital_mute = uda134x_mute,
 		.set_sysclk = uda134x_set_dai_sysclk,
 		.set_fmt = uda134x_set_dai_fmt,
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a69ee72a7af..330877c7069 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -407,7 +407,8 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * when the DAI is being clocked by the CPU DAI. It's up to the
  * machine and cpu DAI driver to do this before we are called.
  */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -439,7 +440,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
 }
 
 static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -477,7 +479,8 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -560,8 +563,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt,
 	},
@@ -579,8 +580,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt,
 	},
@@ -598,8 +597,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.set_fmt = uda1380_set_dai_fmt,
 	},
 },
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index d8ca2da8d63..173b66c0c76 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -463,7 +463,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -585,8 +586,6 @@ struct snd_soc_dai wm8510_dai = {
 		.formats = WM8510_FORMATS,},
 	.ops = {
 		.hw_params = wm8510_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8510_mute,
 		.set_fmt = wm8510_set_dai_fmt,
 		.set_clkdiv = wm8510_set_dai_clkdiv,
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index cbcd7c324ab..220d4b68904 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -548,13 +548,13 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai_link *dai = rtd->dai;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
-	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id);
+	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
 
 	paifb &= ~WM8580_AIF_LENGTH_MASK;
 	/* bit size */
@@ -574,7 +574,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb);
+	wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb);
 	return 0;
 }
 
@@ -798,8 +798,6 @@ struct snd_soc_dai wm8580_dai[] = {
 		},
 		.ops = {
 			 .hw_params = wm8580_paif_hw_params,
-		 },
-		.dai_ops = {
 			 .set_fmt = wm8580_set_paif_dai_fmt,
 			 .set_clkdiv = wm8580_set_dai_clkdiv,
 			 .set_pll = wm8580_set_dai_pll,
@@ -818,8 +816,6 @@ struct snd_soc_dai wm8580_dai[] = {
 		},
 		.ops = {
 			 .hw_params = wm8580_paif_hw_params,
-		 },
-		.dai_ops = {
 			 .set_fmt = wm8580_set_paif_dai_fmt,
 			 .set_clkdiv = wm8580_set_dai_clkdiv,
 			 .set_pll = wm8580_set_dai_pll,
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 3e39dea6124..71949bd320d 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -147,7 +147,8 @@ static int wm8728_mute(struct snd_soc_dai *dai, int mute)
 }
 
 static int wm8728_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -269,8 +270,6 @@ struct snd_soc_dai wm8728_dai = {
 	},
 	.ops = {
 		 .hw_params = wm8728_hw_params,
-	},
-	.dai_ops = {
 		 .digital_mute = wm8728_mute,
 		 .set_fmt = wm8728_set_dai_fmt,
 	}
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7f8a7e36b33..c0f277053bb 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -264,7 +264,8 @@ static inline int get_coeff(int mclk, int rate)
 }
 
 static int wm8731_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -293,7 +294,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -305,7 +307,8 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void wm8731_shutdown(struct snd_pcm_substream *substream)
+static void wm8731_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -461,8 +464,6 @@ struct snd_soc_dai wm8731_dai = {
 		.prepare = wm8731_pcm_prepare,
 		.hw_params = wm8731_hw_params,
 		.shutdown = wm8731_shutdown,
-	},
-	.dai_ops = {
 		.digital_mute = wm8731_mute,
 		.set_sysclk = wm8731_set_dai_sysclk,
 		.set_fmt = wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9b7296ee5b0..860a1d56830 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -614,7 +614,8 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -709,8 +710,6 @@ struct snd_soc_dai wm8750_dai = {
 		.formats = WM8750_FORMATS,},
 	.ops = {
 		.hw_params = wm8750_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8750_mute,
 		.set_fmt = wm8750_set_dai_fmt,
 		.set_sysclk = wm8750_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d426eaa2218..5e4cd3bb824 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -922,7 +922,8 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1155,7 +1156,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1323,16 +1325,15 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS},
 	.capture = { /* dummy for fast DAI switching */
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode1h_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1356,8 +1357,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_pcm_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_pcm_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode1v_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1385,8 +1385,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_pcm_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_pcm_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode2_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1410,8 +1409,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode3_4_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1439,8 +1437,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode3_4_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index de016f41e04..d1326be91c8 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -727,7 +727,8 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int wm8900_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1117,8 +1118,6 @@ struct snd_soc_dai wm8900_dai = {
 	 },
 	.ops = {
 		.hw_params = wm8900_hw_params,
-	 },
-	.dai_ops = {
 		 .set_clkdiv = wm8900_set_dai_clkdiv,
 		 .set_pll = wm8900_set_dai_pll,
 		 .set_fmt = wm8900_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce40d787760..efbe8927b7d 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1257,7 +1257,8 @@ static struct {
 	{ 0,      0 },
 };
 
-static int wm8903_startup(struct snd_pcm_substream *substream)
+static int wm8903_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1298,7 +1299,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void wm8903_shutdown(struct snd_pcm_substream *substream)
+static void wm8903_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1317,7 +1319,8 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream)
 }
 
 static int wm8903_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1515,8 +1518,6 @@ struct snd_soc_dai wm8903_dai = {
 		 .startup = wm8903_startup,
 		 .shutdown = wm8903_shutdown,
 		 .hw_params = wm8903_hw_params,
-	},
-	.dai_ops = {
 		 .digital_mute = wm8903_digital_mute,
 		 .set_fmt = wm8903_set_dai_fmt,
 		 .set_sysclk = wm8903_set_dai_sysclk
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index f41a578ddd4..26edcc9d6e8 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -541,7 +541,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -634,8 +635,6 @@ struct snd_soc_dai wm8971_dai = {
 		.formats = WM8971_FORMATS,},
 	.ops = {
 		.hw_params = wm8971_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8971_mute,
 		.set_fmt = wm8971_set_dai_fmt,
 		.set_sysclk = wm8971_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 2d7b0096d92..13926516d16 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1172,7 +1172,8 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8990_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1362,8 +1363,7 @@ struct snd_soc_dai wm8990_dai = {
 		.rates = WM8990_RATES,
 		.formats = WM8990_FORMATS,},
 	.ops = {
-		.hw_params = wm8990_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8990_hw_params,
 		.digital_mute = wm8990_mute,
 		.set_fmt = wm8990_set_dai_fmt,
 		.set_clkdiv = wm8990_set_dai_clkdiv,
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ffb471e420e..81c38e7ad34 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -487,7 +487,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 	return 0;
 }
 
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -507,7 +508,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
 	return ac97_write(codec, reg, runtime->rate);
 }
 
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 740bf3cde18..a0cc5ac969a 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -928,7 +928,8 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -954,7 +955,8 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
+static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -969,7 +971,8 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
 	ac97_write(codec, AC97_EXTENDED_MID, status);
 }
 
-static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
+static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -989,7 +992,8 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
 	return ac97_write(codec, reg, runtime->rate);
 }
 
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -1042,8 +1046,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
-		.prepare = ac97_hifi_prepare,},
-	.dai_ops = {
+		.prepare = ac97_hifi_prepare,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,},
 	},
@@ -1056,8 +1059,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
-		.prepare = ac97_aux_prepare,},
-	.dai_ops = {
+		.prepare = ac97_aux_prepare,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,},
 	},
@@ -1077,8 +1079,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.formats = WM9713_PCM_FORMATS,},
 	.ops = {
 		.hw_params = wm9713_pcm_hw_params,
-		.shutdown = wm9713_voiceshutdown,},
-	.dai_ops = {
+		.shutdown = wm9713_voiceshutdown,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,
 		.set_fmt = wm9713_set_dai_fmt,
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 95df51e803b..7a17cd0ecf6 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -188,7 +188,8 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream)
 	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
 }
 
-static int davinci_i2s_startup(struct snd_pcm_substream *substream)
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -285,7 +286,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
@@ -349,7 +351,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -473,8 +476,7 @@ struct snd_soc_dai davinci_i2s_dai = {
 	.ops = {
 		.startup = davinci_i2s_startup,
 		.trigger = davinci_i2s_trigger,
-		.hw_params = davinci_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = davinci_i2s_hw_params,
 		.set_fmt = davinci_i2s_set_dai_fmt,
 	},
 };
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index fa38f9cd350..e95fde1766b 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -34,7 +34,8 @@
 #include "davinci-i2s.h"
 
 static int sffsdr_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 157a7895ffa..52c290bb47b 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -266,7 +266,8 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
  * If this is the first stream open, then grab the IRQ and program most of
  * the SSI registers.
  */
-static int fsl_ssi_startup(struct snd_pcm_substream *substream)
+static int fsl_ssi_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -411,7 +412,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
  * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
  * clock master.
  */
-static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
+static int fsl_ssi_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -441,7 +443,8 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
  * The DMA channel is in external master start and pause mode, which
  * means the SSI completely controls the flow of data.
  */
-static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -490,7 +493,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the SSI if there are no other substreams open.
  */
-static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -578,8 +582,6 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.prepare = fsl_ssi_prepare,
 		.shutdown = fsl_ssi_shutdown,
 		.trigger = fsl_ssi_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = fsl_ssi_set_sysclk,
 		.set_fmt = fsl_ssi_set_fmt,
 	},
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 94a02eaa482..e2c172f3897 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -187,7 +187,8 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
  * If this is the first stream open, then grab the IRQ and program most of
  * the PSC registers.
  */
-static int psc_i2s_startup(struct snd_pcm_substream *substream)
+static int psc_i2s_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -220,7 +221,8 @@ static int psc_i2s_startup(struct snd_pcm_substream *substream)
 }
 
 static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -256,7 +258,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
@@ -268,7 +271,8 @@ static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
  * This function is called by ALSA to start, stop, pause, and resume the DMA
  * transfer of data.
  */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -383,7 +387,8 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the PSC if there are no other substreams open.
  */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
+static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -483,8 +488,6 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.hw_free = psc_i2s_hw_free,
 		.shutdown = psc_i2s_shutdown,
 		.trigger = psc_i2s_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = psc_i2s_set_sysclk,
 		.set_fmt = psc_i2s_set_fmt,
 	},
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 3d4060b00eb..6013898f49b 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -138,7 +138,8 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
 static const unsigned long omap34xx_mcbsp_port[][2] = {};
 #endif
 
-static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -151,7 +152,8 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 	return err;
 }
 
-static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -163,7 +165,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
 	}
 }
 
-static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -192,7 +195,8 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
-				    struct snd_pcm_hw_params *params)
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -470,8 +474,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		.shutdown = omap_mcbsp_dai_shutdown,		\
 		.trigger = omap_mcbsp_dai_trigger,		\
 		.hw_params = omap_mcbsp_dai_hw_params,		\
-	},							\
-	.dai_ops = {						\
 		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
 		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
 		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index 5bea31157a9..7b160f9d83f 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -38,7 +38,8 @@
 #include "../codecs/twl4030.h"
 
 static int omap2evm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index e2b54b88c38..d0dd6245a20 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -212,7 +212,8 @@ static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
 	},
 };
 
-static int pxa_ssp_startup(struct snd_pcm_substream *substream)
+static int pxa_ssp_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -228,7 +229,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream)
 	return ret;
 }
 
-static void pxa_ssp_shutdown(struct snd_pcm_substream *substream)
+static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -604,7 +606,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
  * Can be called multiple times by oss emulation.
  */
 static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -678,7 +681,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd)
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -806,8 +810,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.shutdown = pxa_ssp_shutdown,
 			.trigger = pxa_ssp_trigger,
 			.hw_params = pxa_ssp_hw_params,
-		 },
-		.dai_ops = {
 			.set_sysclk = pxa_ssp_set_dai_sysclk,
 			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
 			.set_pll = pxa_ssp_set_dai_pll,
@@ -840,8 +842,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.shutdown = pxa_ssp_shutdown,
 			.trigger = pxa_ssp_trigger,
 			.hw_params = pxa_ssp_hw_params,
-		 },
-		.dai_ops = {
 			.set_sysclk = pxa_ssp_set_dai_sysclk,
 			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
 			.set_pll = pxa_ssp_set_dai_pll,
@@ -875,8 +875,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.shutdown = pxa_ssp_shutdown,
 			.trigger = pxa_ssp_trigger,
 			.hw_params = pxa_ssp_hw_params,
-		 },
-		.dai_ops = {
 			.set_sysclk = pxa_ssp_set_dai_sysclk,
 			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
 			.set_pll = pxa_ssp_set_dai_pll,
@@ -910,8 +908,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.shutdown = pxa_ssp_shutdown,
 			.trigger = pxa_ssp_trigger,
 			.hw_params = pxa_ssp_hw_params,
-		 },
-		.dai_ops = {
 			.set_sysclk = pxa_ssp_set_dai_sysclk,
 			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
 			.set_pll = pxa_ssp_set_dai_pll,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a7a3a9c5c6f..86667d2f1b7 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -117,7 +117,8 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev,
 }
 
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -131,7 +132,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -145,7 +147,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index e758034db5c..9a3e55b4812 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -121,7 +121,8 @@ static struct pxa2xx_gpio gpio_bus[] = {
 	},
 };
 
-static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -187,7 +188,8 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 }
 
 static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -248,7 +250,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			      struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -269,7 +272,8 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
 	return ret;
 }
 
-static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
 {
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		SACR1 |= SACR1_DRPL;
@@ -353,8 +357,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.startup = pxa2xx_i2s_startup,
 		.shutdown = pxa2xx_i2s_shutdown,
 		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = pxa2xx_i2s_hw_params,
 		.set_fmt = pxa2xx_i2s_set_dai_fmt,
 		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
 	},
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ded7d995a92..360cc2a49d9 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -343,7 +343,8 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
@@ -373,7 +374,8 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
@@ -730,8 +732,6 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 	.ops = {
 		.trigger	= s3c2412_i2s_trigger,
 		.hw_params	= s3c2412_i2s_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt	= s3c2412_i2s_set_fmt,
 		.set_clkdiv	= s3c2412_i2s_set_clkdiv,
 		.set_sysclk	= s3c2412_i2s_set_sysclk,
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 19c5c3cf5d8..31377821b2c 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -271,7 +271,8 @@ static void s3c2443_ac97_remove(struct platform_device *pdev,
 }
 
 static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -313,7 +314,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				      struct snd_pcm_hw_params *params,
+				      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -327,7 +329,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 }
 
 static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
-	int cmd)
+				    int cmd, struct snd_soc_dai *dai)
 {
 	u32 ac_glbctrl;
 
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index c18977bceaf..1bac9dd3dbd 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -243,7 +243,8 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
@@ -279,7 +280,8 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -475,8 +477,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
 		.trigger = s3c24xx_i2s_trigger,
-		.hw_params = s3c24xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = s3c24xx_i2s_hw_params,
 		.set_fmt = s3c24xx_i2s_set_fmt,
 		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
 		.set_sysclk = s3c24xx_i2s_set_sysclk,
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index df7bc345c32..3318071dc80 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -236,7 +236,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static int hac_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
+			 struct snd_pcm_hw_params *params,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id];
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 55c3464163a..52a233840d2 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -89,7 +89,8 @@ struct ssi_priv {
  * track usage of the SSI; it is simplex-only so prevent attempts of
  * concurrent playback + capture. FIXME: any locking required?
  */
-static int ssi_startup(struct snd_pcm_substream *substream)
+static int ssi_startup(struct snd_pcm_substream *substream,
+		       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -101,7 +102,8 @@ static int ssi_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void ssi_shutdown(struct snd_pcm_substream *substream)
+static void ssi_shutdown(struct snd_pcm_substream *substream,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -109,7 +111,8 @@ static void ssi_shutdown(struct snd_pcm_substream *substream)
 	ssi->inuse = 0;
 }
 
-static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+		       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -129,7 +132,8 @@ static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int ssi_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
+			 struct snd_pcm_hw_params *params,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -354,8 +358,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.shutdown	= ssi_shutdown,
 		.trigger	= ssi_trigger,
 		.hw_params	= ssi_hw_params,
-	},
-	.dai_ops = {
 		.set_sysclk	= ssi_set_sysclk,
 		.set_clkdiv	= ssi_set_clkdiv,
 		.set_fmt	= ssi_set_fmt,
@@ -383,8 +385,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.shutdown	= ssi_shutdown,
 		.trigger	= ssi_trigger,
 		.hw_params	= ssi_hw_params,
-	},
-	.dai_ops = {
 		.set_sysclk	= ssi_set_sysclk,
 		.set_clkdiv	= ssi_set_clkdiv,
 		.set_fmt	= ssi_set_fmt,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c5cb9516fea..43f4060dbe7 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -134,7 +134,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 
 	/* startup the audio subsystem */
 	if (cpu_dai->ops.startup) {
-		ret = cpu_dai->ops.startup(substream);
+		ret = cpu_dai->ops.startup(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open interface %s\n",
 				cpu_dai->name);
@@ -151,7 +151,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	}
 
 	if (codec_dai->ops.startup) {
-		ret = codec_dai->ops.startup(substream);
+		ret = codec_dai->ops.startup(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open codec %s\n",
 				codec_dai->name);
@@ -248,7 +248,7 @@ codec_dai_err:
 
 platform_err:
 	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream);
+		cpu_dai->ops.shutdown(substream, cpu_dai);
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -339,10 +339,10 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 		snd_soc_dai_digital_mute(codec_dai, 1);
 
 	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream);
+		cpu_dai->ops.shutdown(substream, cpu_dai);
 
 	if (codec_dai->ops.shutdown)
-		codec_dai->ops.shutdown(substream);
+		codec_dai->ops.shutdown(substream, codec_dai);
 
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
@@ -406,7 +406,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	}
 
 	if (codec_dai->ops.prepare) {
-		ret = codec_dai->ops.prepare(substream);
+		ret = codec_dai->ops.prepare(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: codec DAI prepare error\n");
 			goto out;
@@ -414,7 +414,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	}
 
 	if (cpu_dai->ops.prepare) {
-		ret = cpu_dai->ops.prepare(substream);
+		ret = cpu_dai->ops.prepare(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
 			goto out;
@@ -491,7 +491,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (codec_dai->ops.hw_params) {
-		ret = codec_dai->ops.hw_params(substream, params);
+		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
 				codec_dai->name);
@@ -500,7 +500,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (cpu_dai->ops.hw_params) {
-		ret = cpu_dai->ops.hw_params(substream, params);
+		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
@@ -523,11 +523,11 @@ out:
 
 platform_err:
 	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream);
+		cpu_dai->ops.hw_free(substream, cpu_dai);
 
 interface_err:
 	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream);
+		codec_dai->ops.hw_free(substream, codec_dai);
 
 codec_err:
 	if (machine->ops && machine->ops->hw_free)
@@ -566,10 +566,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	/* now free hw params for the DAI's  */
 	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream);
+		codec_dai->ops.hw_free(substream, codec_dai);
 
 	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream);
+		cpu_dai->ops.hw_free(substream, cpu_dai);
 
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -586,7 +586,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	int ret;
 
 	if (codec_dai->ops.trigger) {
-		ret = codec_dai->ops.trigger(substream, cmd);
+		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -598,7 +598,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	}
 
 	if (cpu_dai->ops.trigger) {
-		ret = cpu_dai->ops.trigger(substream, cmd);
+		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -637,10 +637,10 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);
 
 	/* mute any active DAC's */
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
-		if (dai->dai_ops.digital_mute && dai->playback.active)
-			dai->dai_ops.digital_mute(dai, 1);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+		if (dai->ops.digital_mute && dai->playback.active)
+			dai->ops.digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
@@ -733,10 +733,10 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	/* unmute any active DACs */
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
-		if (dai->dai_ops.digital_mute && dai->playback.active)
-			dai->dai_ops.digital_mute(dai, 0);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+		if (dai->ops.digital_mute && dai->playback.active)
+			dai->ops.digital_mute(dai, 0);
 	}
 
 	for (i = 0; i < card->num_links; i++) {
@@ -1849,8 +1849,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
 }
@@ -1869,8 +1869,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->dai_ops.set_clkdiv)
-		return dai->dai_ops.set_clkdiv(dai, div_id, div);
+	if (dai->ops.set_clkdiv)
+		return dai->ops.set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
 }
@@ -1888,8 +1888,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->dai_ops.set_pll)
-		return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out);
+	if (dai->ops.set_pll)
+		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -1905,8 +1905,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->dai_ops.set_fmt)
-		return dai->dai_ops.set_fmt(dai, fmt);
+	if (dai->ops.set_fmt)
+		return dai->ops.set_fmt(dai, fmt);
 	else
 		return -EINVAL;
 }
@@ -1924,8 +1924,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_tdm_slot(dai, mask, slots);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
 }
@@ -1940,8 +1940,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_tristate(dai, tristate);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_tristate(dai, tristate);
 	else
 		return -EINVAL;
 }
@@ -1956,8 +1956,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->dai_ops.digital_mute)
-		return dai->dai_ops.digital_mute(dai, mute);
+	if (dai->ops.digital_mute)
+		return dai->ops.digital_mute(dai, mute);
 	else
 		return -EINVAL;
 }
-- 
cgit v1.2.3-70-g09d2


From 2dac9217b26fd0a0a1712386ce2ea1411835ffb7 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 21 Nov 2008 14:01:41 +0000
Subject: ASoC: Add Marvell Zylonite machine support

Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.

The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Makefile   |   2 +
 sound/soc/pxa/zylonite.c | 219 +++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 221 insertions(+)
 create mode 100644 sound/soc/pxa/zylonite.c

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 1a3a36e75bf..08a9f279772 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -17,6 +17,7 @@ snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
+snd-soc-zylonite-objs := zylonite.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -25,3 +26,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
 obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
new file mode 100644
index 00000000000..842d6500d61
--- /dev/null
+++ b/sound/soc/pxa/zylonite.c
@@ -0,0 +1,219 @@
+/*
+ * zylonite.c  --  SoC audio for Zylonite
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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; either version 2 of the
+ * License, or (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm9713.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "pxa-ssp.h"
+
+static struct snd_soc_card zylonite;
+
+static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Microphone", NULL),
+	SND_SOC_DAPM_MIC("Handset Microphone", NULL),
+	SND_SOC_DAPM_SPK("Multiactor", NULL),
+	SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
+};
+
+/* Currently supported audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Headphone output connected to HPL/HPR */
+	{ "Headphone", NULL,  "HPL" },
+	{ "Headphone", NULL,  "HPR" },
+
+	/* On-board earpiece */
+	{ "Headset Earpiece", NULL, "OUT3" },
+
+	/* Headphone mic */
+	{ "MIC2A", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Headset Microphone" },
+
+	/* On-board mic */
+	{ "MIC1", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Handset Microphone" },
+
+	/* Multiactor differentially connected over SPKL/SPKR */
+	{ "Multiactor", NULL, "SPKL" },
+	{ "Multiactor", NULL, "SPKR" },
+};
+
+static int zylonite_wm9713_init(struct snd_soc_codec *codec)
+{
+	/* Currently we only support use of the AC97 clock here.  If
+	 * CLK_POUT is selected by SW15 then the clock API will need
+	 * to be used to request and enable it here.
+	 */
+
+	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
+				  ARRAY_SIZE(zylonite_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* Static setup for now */
+	snd_soc_dapm_enable_pin(codec, "Headphone");
+	snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static int zylonite_voice_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;
+	unsigned int pll_out = 0;
+	unsigned int acds = 0;
+	unsigned int wm9713_div = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+		wm9713_div = 12;
+		pll_out = 2048000;
+		break;
+	case 16000:
+		wm9713_div = 6;
+		pll_out = 4096000;
+		break;
+	case 48000:
+	default:
+		wm9713_div = 2;
+		pll_out = 12288000;
+		acds = 1;
+		break;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai,
+				       params_channels(params),
+				       params_channels(params));
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+	if (ret < 0)
+		return ret;
+
+	/* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
+	 * to be set instead.
+	 */
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+				     WM9713_PCMDIV(wm9713_div));
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops zylonite_voice_ops = {
+	.hw_params = zylonite_voice_hw_params,
+};
+
+static struct snd_soc_dai_link zylonite_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+	.init = zylonite_wm9713_init,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+},
+{
+	.name = "WM9713 Voice",
+	.stream_name = "WM9713 Voice",
+	.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3],
+	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
+	.ops = &zylonite_voice_ops,
+},
+};
+
+static struct snd_soc_card zylonite = {
+	.name = "Zylonite",
+	.dai_link = zylonite_dai,
+	.num_links = ARRAY_SIZE(zylonite_dai),
+};
+
+static struct snd_soc_device zylonite_snd_ac97_devdata = {
+	.card = &zylonite,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm9713,
+};
+
+static struct platform_device *zylonite_snd_ac97_device;
+
+static int __init zylonite_init(void)
+{
+	int ret;
+
+	zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+	if (!zylonite_snd_ac97_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(zylonite_snd_ac97_device,
+			     &zylonite_snd_ac97_devdata);
+	zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev;
+
+	ret = platform_device_add(zylonite_snd_ac97_device);
+	if (ret != 0)
+		platform_device_put(zylonite_snd_ac97_device);
+
+	return ret;
+}
+
+static void __exit zylonite_exit(void)
+{
+	platform_device_unregister(zylonite_snd_ac97_device);
+}
+
+module_init(zylonite_init);
+module_exit(zylonite_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3-70-g09d2


From 0e734ad5d16ad1d87a428a30d117bb3541a8e24d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 21 Nov 2008 14:05:48 +0000
Subject: ASoC: Staticise pxa2xx_pcm_ops

It's not exported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/pxa2xx-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index afcd892cd2f..0f6b7bb2d44 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -69,7 +69,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-struct snd_pcm_ops pxa2xx_pcm_ops = {
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
 	.open		= __pxa2xx_pcm_open,
 	.close		= __pxa2xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3-70-g09d2


From 3ba9e10a6d3b6abf5f5952572cff8f8d5a35ae54 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 24 Nov 2008 18:01:05 +0000
Subject: ASoC: Remove DAI type information

DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h            |  2 +-
 include/sound/soc.h                |  8 --------
 sound/soc/atmel/atmel_ssc_dai.c    |  3 ---
 sound/soc/au1x/psc-ac97.c          |  2 +-
 sound/soc/au1x/psc-i2s.c           |  1 -
 sound/soc/blackfin/bf5xx-ac97.c    |  2 +-
 sound/soc/blackfin/bf5xx-i2s.c     |  1 -
 sound/soc/codecs/ac97.c            |  2 +-
 sound/soc/codecs/pcm3008.c         |  1 -
 sound/soc/codecs/wm9712.c          |  2 +-
 sound/soc/codecs/wm9713.c          |  2 +-
 sound/soc/davinci/davinci-i2s.c    |  1 -
 sound/soc/fsl/mpc5200_psc_i2s.c    |  1 -
 sound/soc/omap/omap-mcbsp.c        |  1 -
 sound/soc/pxa/pxa-ssp.c            |  4 ----
 sound/soc/pxa/pxa2xx-ac97.c        |  6 +++---
 sound/soc/pxa/pxa2xx-i2s.c         |  1 -
 sound/soc/s3c24xx/neo1973_wm8753.c |  1 -
 sound/soc/s3c24xx/s3c2412-i2s.c    |  1 -
 sound/soc/s3c24xx/s3c2443-ac97.c   |  4 ++--
 sound/soc/s3c24xx/s3c24xx-i2s.c    |  1 -
 sound/soc/sh/hac.c                 |  4 ++--
 sound/soc/sh/ssi.c                 |  2 --
 sound/soc/soc-core.c               | 31 ++++++++-----------------------
 24 files changed, 21 insertions(+), 63 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index f51cb55902f..a01a24b1019 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -184,7 +184,7 @@ struct snd_soc_dai {
 	/* DAI description */
 	char *name;
 	unsigned int id;
-	unsigned char type;
+	int ac97_control;
 
 	/* DAI callbacks */
 	int (*probe)(struct platform_device *pdev,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index e4465f73aa4..444f9c21137 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -143,14 +143,6 @@ enum snd_soc_bias_level {
 	SND_SOC_BIAS_OFF,
 };
 
-/*
- * Digital Audio Interface (DAI) types
- */
-#define SND_SOC_DAI_AC97	0x1
-#define SND_SOC_DAI_I2S		0x2
-#define SND_SOC_DAI_PCM		0x4
-#define SND_SOC_DAI_AC97_BUS	0x8	/* for custom i.e. non ac97_codec.c */
-
 struct snd_soc_device;
 struct snd_soc_pcm_stream;
 struct snd_soc_ops;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 916f73b9a18..0bb18dfa949 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -702,7 +702,6 @@ static int atmel_ssc_resume(struct platform_device *pdev,
 struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 	{	.name = "atmel-ssc0",
 		.id = 0,
-		.type = SND_SOC_DAI_PCM,
 		.suspend = atmel_ssc_suspend,
 		.resume = atmel_ssc_resume,
 		.playback = {
@@ -727,7 +726,6 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 #if NUM_SSC_DEVICES == 3
 	{	.name = "atmel-ssc1",
 		.id = 1,
-		.type = SND_SOC_DAI_PCM,
 		.suspend = atmel_ssc_suspend,
 		.resume = atmel_ssc_resume,
 		.playback = {
@@ -751,7 +749,6 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 	},
 	{	.name = "atmel-ssc2",
 		.id = 2,
-		.type = SND_SOC_DAI_PCM,
 		.suspend = atmel_ssc_suspend,
 		.resume = atmel_ssc_resume,
 		.playback = {
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index ad60a6042ca..a0bcfeaf5f8 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -346,7 +346,7 @@ static int au1xpsc_ac97_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
-	.type			= SND_SOC_DAI_AC97,
+	.ac97_control		= 1,
 	.probe			= au1xpsc_ac97_probe,
 	.remove			= au1xpsc_ac97_remove,
 	.suspend		= au1xpsc_ac97_suspend,
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 05a5acbb16a..f4217e70a78 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -371,7 +371,6 @@ static int au1xpsc_i2s_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
-	.type			= SND_SOC_DAI_I2S,
 	.probe			= au1xpsc_i2s_probe,
 	.remove			= au1xpsc_i2s_remove,
 	.suspend		= au1xpsc_i2s_suspend,
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 5dcd3f665ab..709bdf08e39 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -409,7 +409,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
 struct snd_soc_dai bfin_ac97_dai = {
 	.name = "bf5xx-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = bf5xx_ac97_probe,
 	.remove = bf5xx_ac97_remove,
 	.suspend = bf5xx_ac97_suspend,
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 4e675b55b18..6e5036bf924 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -292,7 +292,6 @@ static int bf5xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai bf5xx_i2s_dai = {
 	.name = "bf5xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = bf5xx_i2s_probe,
 	.remove = bf5xx_i2s_remove,
 	.suspend = bf5xx_i2s_suspend,
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 8a93aff359d..c4208c8210c 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -43,7 +43,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 
 struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 1,
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index 2b26e1d80c8..651a15eb8c2 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -33,7 +33,6 @@
 
 struct snd_soc_dai pcm3008_dai = {
 	.name = "PCM3008 HiFi",
-	.type = SND_SOC_DAI_I2S,
 	.playback = {
 		.stream_name = "PCM3008 Playback",
 		.channels_min = 1,
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 6e3e0f34017..40f14061fb7 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -535,7 +535,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97_BUS,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "HiFi Playback",
 		.channels_min = 1,
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index a502667fca7..9dad0bffcb0 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1024,7 +1024,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97_BUS,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "HiFi Playback",
 		.channels_min = 1,
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 7a17cd0ecf6..cf31b3bb96c 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -460,7 +460,6 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = davinci_i2s_probe,
 	.remove = davinci_i2s_remove,
 	.playback = {
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index e2c172f3897..9ad8f9a2d8e 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -469,7 +469,6 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
 static struct snd_soc_dai psc_i2s_dai_template = {
-	.type = SND_SOC_DAI_I2S,
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 2eeb135c1e4..252bc7ebb19 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -456,7 +456,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 {								\
 	.name = "omap-mcbsp-dai-"#link_id,			\
 	.id = (link_id),					\
-	.type = SND_SOC_DAI_I2S,				\
 	.playback = {						\
 		.channels_min = 2,				\
 		.channels_max = 2,				\
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d0dd6245a20..402fc5ba65e 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -788,7 +788,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 	{
 		.name = "pxa2xx-ssp1",
 		.id = 0,
-		.type = SND_SOC_DAI_PCM,
 		.probe = pxa_ssp_probe,
 		.remove = pxa_ssp_remove,
 		.suspend = pxa_ssp_suspend,
@@ -820,7 +819,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 	},
 	{	.name = "pxa2xx-ssp2",
 		.id = 1,
-		.type = SND_SOC_DAI_PCM,
 		.probe = pxa_ssp_probe,
 		.remove = pxa_ssp_remove,
 		.suspend = pxa_ssp_suspend,
@@ -853,7 +851,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 	{
 		.name = "pxa2xx-ssp3",
 		.id = 2,
-		.type = SND_SOC_DAI_PCM,
 		.probe = pxa_ssp_probe,
 		.remove = pxa_ssp_remove,
 		.suspend = pxa_ssp_suspend,
@@ -886,7 +883,6 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 	{
 		.name = "pxa2xx-ssp4",
 		.id = 3,
-		.type = SND_SOC_DAI_PCM,
 		.probe = pxa_ssp_probe,
 		.remove = pxa_ssp_remove,
 		.suspend = pxa_ssp_suspend,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 86667d2f1b7..bffbe288634 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -173,7 +173,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = pxa2xx_ac97_probe,
 	.remove = pxa2xx_ac97_remove,
 	.suspend = pxa2xx_ac97_suspend,
@@ -196,7 +196,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-aux",
 	.id = 1,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Aux Playback",
 		.channels_min = 1,
@@ -215,7 +215,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-mic",
 	.id = 2,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
 		.channels_min = 1,
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 9a3e55b4812..f9a9e2ebafa 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -340,7 +340,6 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.suspend = pxa2xx_i2s_suspend,
 	.resume = pxa2xx_i2s_resume,
 	.playback = {
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 528fc3f1b45..3df2224a672 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -548,7 +548,6 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai bt_dai = {
 	.name = "Bluetooth",
 	.id = 0,
-	.type = SND_SOC_DAI_PCM,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 1,
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 360cc2a49d9..1c741047ae3 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -713,7 +713,6 @@ static int s3c2412_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai s3c2412_i2s_dai = {
 	.name	= "s3c2412-i2s",
 	.id	= 0,
-	.type	= SND_SOC_DAI_I2S,
 	.probe	= s3c2412_i2s_probe,
 	.suspend = s3c2412_i2s_suspend,
 	.resume = s3c2412_i2s_resume,
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 31377821b2c..41bde6a3883 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -358,7 +358,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = s3c2443_ac97_probe,
 	.remove = s3c2443_ac97_remove,
 	.playback = {
@@ -380,7 +380,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-mic",
 	.id = 1,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
 		.channels_min = 1,
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 1bac9dd3dbd..8d9135f41bc 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -461,7 +461,6 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = s3c24xx_i2s_probe,
 	.suspend = s3c24xx_i2s_suspend,
 	.resume = s3c24xx_i2s_resume,
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index 3318071dc80..c435913c60e 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -271,7 +271,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 {
 	.name			= "HAC0",
 	.id			= 0,
-	.type			= SND_SOC_DAI_AC97,
+	.ac97_control		= 1,
 	.playback = {
 		.rates		= AC97_RATES,
 		.formats	= AC97_FMTS,
@@ -291,8 +291,8 @@ struct snd_soc_dai sh4_hac_dai[] = {
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
 	.name			= "HAC1",
+	.ac97_control		= 1,
 	.id			= 1,
-	.type			= SND_SOC_DAI_AC97,
 	.playback = {
 		.rates		= AC97_RATES,
 		.formats	= AC97_FMTS,
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 52a233840d2..fed544a3def 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -340,7 +340,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
 	.id			= 0,
-	.type			= SND_SOC_DAI_I2S,
 	.playback = {
 		.rates		= SSI_RATES,
 		.formats	= SSI_FMTS,
@@ -367,7 +366,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI1",
 	.id			= 1,
-	.type			= SND_SOC_DAI_I2S,
 	.playback = {
 		.rates		= SSI_RATES,
 		.formats	= SSI_FMTS,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 43f4060dbe7..0d47696ccd0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -100,20 +100,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
-static inline const char *get_dai_name(int type)
-{
-	switch (type) {
-	case SND_SOC_DAI_AC97_BUS:
-	case SND_SOC_DAI_AC97:
-		return "AC97";
-	case SND_SOC_DAI_I2S:
-		return "I2S";
-	case SND_SOC_DAI_PCM:
-		return "PCM";
-	}
-	return NULL;
-}
-
 /*
  * Called by ALSA when a PCM substream is opened, the runtime->hw record is
  * then initialized and any private data can be allocated. This also calls
@@ -652,7 +638,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
-		if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
+		if (cpu_dai->suspend && !cpu_dai->ac97_control)
 			cpu_dai->suspend(pdev, cpu_dai);
 		if (platform->suspend)
 			platform->suspend(pdev, cpu_dai);
@@ -678,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
-		if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
+		if (cpu_dai->suspend && cpu_dai->ac97_control)
 			cpu_dai->suspend(pdev, cpu_dai);
 	}
 
@@ -714,7 +700,7 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
-		if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
+		if (cpu_dai->resume && cpu_dai->ac97_control)
 			cpu_dai->resume(pdev, cpu_dai);
 	}
 
@@ -741,7 +727,7 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
-		if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
+		if (cpu_dai->resume && !cpu_dai->ac97_control)
 			cpu_dai->resume(pdev, cpu_dai);
 		if (platform->resume)
 			platform->resume(pdev, cpu_dai);
@@ -898,8 +884,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	codec_dai->codec = socdev->codec;
 
 	/* check client and interface hw capabilities */
-	sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
-		get_dai_name(cpu_dai->type), num);
+	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
+		num);
 
 	if (codec_dai->playback.channels_min)
 		playback = 1;
@@ -1270,8 +1256,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 				continue;
 			}
 		}
-		if (card->dai_link[i].codec_dai->type ==
-			SND_SOC_DAI_AC97_BUS)
+		if (card->dai_link[i].codec_dai->ac97_control)
 			ac97 = 1;
 	}
 	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
@@ -1335,7 +1320,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev)
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
-		if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
+		if (codec_dai->ac97_control && codec->ac97) {
 			soc_ac97_dev_unregister(codec);
 			goto free_card;
 		}
-- 
cgit v1.2.3-70-g09d2


From 5c0d7bb797a975691ca8bbc38e53da03c6e151bb Mon Sep 17 00:00:00 2001
From: Dmitry Baryshkov <dbaryshkov@gmail.com>
Date: Tue, 25 Nov 2008 09:35:21 +0300
Subject: ASoC: tosa: move gpio probing to machine callbacks

Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/tosa.c | 30 ++++++++++++++++++++++--------
 1 file changed, 22 insertions(+), 8 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 9d9be5a14d1..48242b32a28 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -230,10 +230,32 @@ static struct snd_soc_dai_link tosa_dai[] = {
 },
 };
 
+static int tosa_probe(struct platform_device *dev)
+{
+	int ret;
+
+	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
+	if (ret)
+		gpio_free(TOSA_GPIO_L_MUTE);
+
+	return ret;
+}
+
+static int tosa_remove(struct platform_device *dev)
+{
+	gpio_free(TOSA_GPIO_L_MUTE);
+	return 0;
+}
+
 static struct snd_soc_card tosa = {
 	.name = "Tosa",
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
+	.probe = tosa_probe,
+	.remove = tosa_remove,
 };
 
 static struct snd_soc_device tosa_snd_devdata = {
@@ -251,11 +273,6 @@ static int __init tosa_init(void)
 	if (!machine_is_tosa())
 		return -ENODEV;
 
-	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
-	if (ret)
-		return ret;
-	gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
-
 	tosa_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!tosa_snd_device) {
 		ret = -ENOMEM;
@@ -272,15 +289,12 @@ static int __init tosa_init(void)
 	platform_device_put(tosa_snd_device);
 
 err_alloc:
-	gpio_free(TOSA_GPIO_L_MUTE);
-
 	return ret;
 }
 
 static void __exit tosa_exit(void)
 {
 	platform_device_unregister(tosa_snd_device);
-	gpio_free(TOSA_GPIO_L_MUTE);
 }
 
 module_init(tosa_init);
-- 
cgit v1.2.3-70-g09d2


From f5d4c67e41a262f0cdfaec1bb0fa8e5952187ef9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 1 Dec 2008 16:29:47 +0100
Subject: ALSA: soc - Remove obsoleted sound/driver.h inclusion

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/pxa/em-x270.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 4a61925c310..d6884b755d5 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -23,7 +23,6 @@
 #include <linux/moduleparam.h>
 #include <linux/device.h>
 
-#include <sound/driver.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
-- 
cgit v1.2.3-70-g09d2


From 5220ed6b321639d68a66bad2082456c1b273f3ea Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 1 Dec 2008 20:00:47 +0100
Subject: ALSA: ASoC: Fix compile warnings on corgi.c

Fix the wrong shutdown callback type.  Also removed the unused variables
there:

  sound/soc/pxa/corgi.c: In function 'corgi_shutdown':
  sound/soc/pxa/corgi.c:114: warning: unused variable 'codec'
  sound/soc/pxa/corgi.c: At top level:
  sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type

Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/pxa/corgi.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 647f056a3cb..e56bf4b6c2a 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -108,15 +108,11 @@ static int corgi_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on corgi */
-static int corgi_shutdown(struct snd_pcm_substream *substream)
+static void corgi_shutdown(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
-
 	/* set = unmute headphone */
 	gpio_set_value(CORGI_GPIO_MUTE_L, 1);
 	gpio_set_value(CORGI_GPIO_MUTE_R, 1);
-	return 0;
 }
 
 static int corgi_hw_params(struct snd_pcm_substream *substream,
-- 
cgit v1.2.3-70-g09d2


From 1f017a9964c5b3b9581d3a5732110cb1e0444281 Mon Sep 17 00:00:00 2001
From: Eric Miao <eric.miao@marvell.com>
Date: Fri, 28 Nov 2008 14:19:33 +0800
Subject: [ARM] pxa: move AC97 register definitions into dedicated regs-ac97.h

The optimal change would be to move the AC97 register definitions into
the AC97 driver, unfortunately, the registers are shared between several
files. Move them into a dedicated regs-ac97.h first.

Signed-off-by: Eric Miao <eric.miao@marvell.com>
---
 arch/arm/mach-pxa/include/mach/pxa-regs.h    | 96 ---------------------------
 arch/arm/mach-pxa/include/mach/regs-ac97.h   | 99 ++++++++++++++++++++++++++++
 drivers/input/touchscreen/mainstone-wm97xx.c |  2 +-
 sound/arm/pxa2xx-ac97-lib.c                  |  2 +-
 sound/arm/pxa2xx-ac97.c                      |  1 +
 sound/soc/pxa/pxa2xx-ac97.c                  |  1 +
 6 files changed, 103 insertions(+), 98 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/regs-ac97.h

(limited to 'sound/soc/pxa')

diff --git a/arch/arm/mach-pxa/include/mach/pxa-regs.h b/arch/arm/mach-pxa/include/mach/pxa-regs.h
index cb9b46de97c..a5650289803 100644
--- a/arch/arm/mach-pxa/include/mach/pxa-regs.h
+++ b/arch/arm/mach-pxa/include/mach/pxa-regs.h
@@ -131,102 +131,6 @@
  * Serial Audio Controller - moved into sound/soc/pxa/pxa2xx-i2s.c
  */
 
-/*
- * AC97 Controller registers
- */
-
-#define POCR		__REG(0x40500000)  /* PCM Out Control Register */
-#define POCR_FEIE	(1 << 3)	/* FIFO Error Interrupt Enable */
-#define POCR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
-
-#define PICR		__REG(0x40500004)  /* PCM In Control Register */
-#define PICR_FEIE	(1 << 3)	/* FIFO Error Interrupt Enable */
-#define PICR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
-
-#define MCCR		__REG(0x40500008)  /* Mic In Control Register */
-#define MCCR_FEIE	(1 << 3)	/* FIFO Error Interrupt Enable */
-#define MCCR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
-
-#define GCR		__REG(0x4050000C)  /* Global Control Register */
-#ifdef CONFIG_PXA3xx
-#define GCR_CLKBPB	(1 << 31)	/* Internal clock enable */
-#endif
-#define GCR_nDMAEN	(1 << 24)	/* non DMA Enable */
-#define GCR_CDONE_IE	(1 << 19)	/* Command Done Interrupt Enable */
-#define GCR_SDONE_IE	(1 << 18)	/* Status Done Interrupt Enable */
-#define GCR_SECRDY_IEN	(1 << 9)	/* Secondary Ready Interrupt Enable */
-#define GCR_PRIRDY_IEN	(1 << 8)	/* Primary Ready Interrupt Enable */
-#define GCR_SECRES_IEN	(1 << 5)	/* Secondary Resume Interrupt Enable */
-#define GCR_PRIRES_IEN	(1 << 4)	/* Primary Resume Interrupt Enable */
-#define GCR_ACLINK_OFF	(1 << 3)	/* AC-link Shut Off */
-#define GCR_WARM_RST	(1 << 2)	/* AC97 Warm Reset */
-#define GCR_COLD_RST	(1 << 1)	/* AC'97 Cold Reset (0 = active) */
-#define GCR_GIE		(1 << 0)	/* Codec GPI Interrupt Enable */
-
-#define POSR		__REG(0x40500010)  /* PCM Out Status Register */
-#define POSR_FIFOE	(1 << 4)	/* FIFO error */
-#define POSR_FSR	(1 << 2)	/* FIFO Service Request */
-
-#define PISR		__REG(0x40500014)  /* PCM In Status Register */
-#define PISR_FIFOE	(1 << 4)	/* FIFO error */
-#define PISR_EOC	(1 << 3)	/* DMA End-of-Chain (exclusive clear) */
-#define PISR_FSR	(1 << 2)	/* FIFO Service Request */
-
-#define MCSR		__REG(0x40500018)  /* Mic In Status Register */
-#define MCSR_FIFOE	(1 << 4)	/* FIFO error */
-#define MCSR_EOC	(1 << 3)	/* DMA End-of-Chain (exclusive clear) */
-#define MCSR_FSR	(1 << 2)	/* FIFO Service Request */
-
-#define GSR		__REG(0x4050001C)  /* Global Status Register */
-#define GSR_CDONE	(1 << 19)	/* Command Done */
-#define GSR_SDONE	(1 << 18)	/* Status Done */
-#define GSR_RDCS	(1 << 15)	/* Read Completion Status */
-#define GSR_BIT3SLT12	(1 << 14)	/* Bit 3 of slot 12 */
-#define GSR_BIT2SLT12	(1 << 13)	/* Bit 2 of slot 12 */
-#define GSR_BIT1SLT12	(1 << 12)	/* Bit 1 of slot 12 */
-#define GSR_SECRES	(1 << 11)	/* Secondary Resume Interrupt */
-#define GSR_PRIRES	(1 << 10)	/* Primary Resume Interrupt */
-#define GSR_SCR		(1 << 9)	/* Secondary Codec Ready */
-#define GSR_PCR		(1 << 8)	/*  Primary Codec Ready */
-#define GSR_MCINT	(1 << 7)	/* Mic In Interrupt */
-#define GSR_POINT	(1 << 6)	/* PCM Out Interrupt */
-#define GSR_PIINT	(1 << 5)	/* PCM In Interrupt */
-#define GSR_ACOFFD	(1 << 3)	/* AC-link Shut Off Done */
-#define GSR_MOINT	(1 << 2)	/* Modem Out Interrupt */
-#define GSR_MIINT	(1 << 1)	/* Modem In Interrupt */
-#define GSR_GSCI	(1 << 0)	/* Codec GPI Status Change Interrupt */
-
-#define CAR		__REG(0x40500020)  /* CODEC Access Register */
-#define CAR_CAIP	(1 << 0)	/* Codec Access In Progress */
-
-#define PCDR		__REG(0x40500040)  /* PCM FIFO Data Register */
-#define MCDR		__REG(0x40500060)  /* Mic-in FIFO Data Register */
-
-#define MOCR		__REG(0x40500100)  /* Modem Out Control Register */
-#define MOCR_FEIE	(1 << 3)	/* FIFO Error */
-#define MOCR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
-
-#define MICR		__REG(0x40500108)  /* Modem In Control Register */
-#define MICR_FEIE	(1 << 3)	/* FIFO Error */
-#define MICR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
-
-#define MOSR		__REG(0x40500110)  /* Modem Out Status Register */
-#define MOSR_FIFOE	(1 << 4)	/* FIFO error */
-#define MOSR_FSR	(1 << 2)	/* FIFO Service Request */
-
-#define MISR		__REG(0x40500118)  /* Modem In Status Register */
-#define MISR_FIFOE	(1 << 4)	/* FIFO error */
-#define MISR_EOC	(1 << 3)	/* DMA End-of-Chain (exclusive clear) */
-#define MISR_FSR	(1 << 2)	/* FIFO Service Request */
-
-#define MODR		__REG(0x40500140)  /* Modem FIFO Data Register */
-
-#define PAC_REG_BASE	__REG(0x40500200)  /* Primary Audio Codec */
-#define SAC_REG_BASE	__REG(0x40500300)  /* Secondary Audio Codec */
-#define PMC_REG_BASE	__REG(0x40500400)  /* Primary Modem Codec */
-#define SMC_REG_BASE	__REG(0x40500500)  /* Secondary Modem Codec */
-
-
 /*
  * Fast Infrared Communication Port - moved into drivers/net/irda/pxaficp_ir.c
  */
diff --git a/arch/arm/mach-pxa/include/mach/regs-ac97.h b/arch/arm/mach-pxa/include/mach/regs-ac97.h
new file mode 100644
index 00000000000..e41b9d202b8
--- /dev/null
+++ b/arch/arm/mach-pxa/include/mach/regs-ac97.h
@@ -0,0 +1,99 @@
+#ifndef __ASM_ARCH_REGS_AC97_H
+#define __ASM_ARCH_REGS_AC97_H
+
+/*
+ * AC97 Controller registers
+ */
+
+#define POCR		__REG(0x40500000)  /* PCM Out Control Register */
+#define POCR_FEIE	(1 << 3)	/* FIFO Error Interrupt Enable */
+#define POCR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
+
+#define PICR		__REG(0x40500004)  /* PCM In Control Register */
+#define PICR_FEIE	(1 << 3)	/* FIFO Error Interrupt Enable */
+#define PICR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
+
+#define MCCR		__REG(0x40500008)  /* Mic In Control Register */
+#define MCCR_FEIE	(1 << 3)	/* FIFO Error Interrupt Enable */
+#define MCCR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
+
+#define GCR		__REG(0x4050000C)  /* Global Control Register */
+#ifdef CONFIG_PXA3xx
+#define GCR_CLKBPB	(1 << 31)	/* Internal clock enable */
+#endif
+#define GCR_nDMAEN	(1 << 24)	/* non DMA Enable */
+#define GCR_CDONE_IE	(1 << 19)	/* Command Done Interrupt Enable */
+#define GCR_SDONE_IE	(1 << 18)	/* Status Done Interrupt Enable */
+#define GCR_SECRDY_IEN	(1 << 9)	/* Secondary Ready Interrupt Enable */
+#define GCR_PRIRDY_IEN	(1 << 8)	/* Primary Ready Interrupt Enable */
+#define GCR_SECRES_IEN	(1 << 5)	/* Secondary Resume Interrupt Enable */
+#define GCR_PRIRES_IEN	(1 << 4)	/* Primary Resume Interrupt Enable */
+#define GCR_ACLINK_OFF	(1 << 3)	/* AC-link Shut Off */
+#define GCR_WARM_RST	(1 << 2)	/* AC97 Warm Reset */
+#define GCR_COLD_RST	(1 << 1)	/* AC'97 Cold Reset (0 = active) */
+#define GCR_GIE		(1 << 0)	/* Codec GPI Interrupt Enable */
+
+#define POSR		__REG(0x40500010)  /* PCM Out Status Register */
+#define POSR_FIFOE	(1 << 4)	/* FIFO error */
+#define POSR_FSR	(1 << 2)	/* FIFO Service Request */
+
+#define PISR		__REG(0x40500014)  /* PCM In Status Register */
+#define PISR_FIFOE	(1 << 4)	/* FIFO error */
+#define PISR_EOC	(1 << 3)	/* DMA End-of-Chain (exclusive clear) */
+#define PISR_FSR	(1 << 2)	/* FIFO Service Request */
+
+#define MCSR		__REG(0x40500018)  /* Mic In Status Register */
+#define MCSR_FIFOE	(1 << 4)	/* FIFO error */
+#define MCSR_EOC	(1 << 3)	/* DMA End-of-Chain (exclusive clear) */
+#define MCSR_FSR	(1 << 2)	/* FIFO Service Request */
+
+#define GSR		__REG(0x4050001C)  /* Global Status Register */
+#define GSR_CDONE	(1 << 19)	/* Command Done */
+#define GSR_SDONE	(1 << 18)	/* Status Done */
+#define GSR_RDCS	(1 << 15)	/* Read Completion Status */
+#define GSR_BIT3SLT12	(1 << 14)	/* Bit 3 of slot 12 */
+#define GSR_BIT2SLT12	(1 << 13)	/* Bit 2 of slot 12 */
+#define GSR_BIT1SLT12	(1 << 12)	/* Bit 1 of slot 12 */
+#define GSR_SECRES	(1 << 11)	/* Secondary Resume Interrupt */
+#define GSR_PRIRES	(1 << 10)	/* Primary Resume Interrupt */
+#define GSR_SCR		(1 << 9)	/* Secondary Codec Ready */
+#define GSR_PCR		(1 << 8)	/*  Primary Codec Ready */
+#define GSR_MCINT	(1 << 7)	/* Mic In Interrupt */
+#define GSR_POINT	(1 << 6)	/* PCM Out Interrupt */
+#define GSR_PIINT	(1 << 5)	/* PCM In Interrupt */
+#define GSR_ACOFFD	(1 << 3)	/* AC-link Shut Off Done */
+#define GSR_MOINT	(1 << 2)	/* Modem Out Interrupt */
+#define GSR_MIINT	(1 << 1)	/* Modem In Interrupt */
+#define GSR_GSCI	(1 << 0)	/* Codec GPI Status Change Interrupt */
+
+#define CAR		__REG(0x40500020)  /* CODEC Access Register */
+#define CAR_CAIP	(1 << 0)	/* Codec Access In Progress */
+
+#define PCDR		__REG(0x40500040)  /* PCM FIFO Data Register */
+#define MCDR		__REG(0x40500060)  /* Mic-in FIFO Data Register */
+
+#define MOCR		__REG(0x40500100)  /* Modem Out Control Register */
+#define MOCR_FEIE	(1 << 3)	/* FIFO Error */
+#define MOCR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
+
+#define MICR		__REG(0x40500108)  /* Modem In Control Register */
+#define MICR_FEIE	(1 << 3)	/* FIFO Error */
+#define MICR_FSRIE	(1 << 1)	/* FIFO Service Request Interrupt Enable */
+
+#define MOSR		__REG(0x40500110)  /* Modem Out Status Register */
+#define MOSR_FIFOE	(1 << 4)	/* FIFO error */
+#define MOSR_FSR	(1 << 2)	/* FIFO Service Request */
+
+#define MISR		__REG(0x40500118)  /* Modem In Status Register */
+#define MISR_FIFOE	(1 << 4)	/* FIFO error */
+#define MISR_EOC	(1 << 3)	/* DMA End-of-Chain (exclusive clear) */
+#define MISR_FSR	(1 << 2)	/* FIFO Service Request */
+
+#define MODR		__REG(0x40500140)  /* Modem FIFO Data Register */
+
+#define PAC_REG_BASE	__REG(0x40500200)  /* Primary Audio Codec */
+#define SAC_REG_BASE	__REG(0x40500300)  /* Secondary Audio Codec */
+#define PMC_REG_BASE	__REG(0x40500400)  /* Primary Modem Codec */
+#define SMC_REG_BASE	__REG(0x40500500)  /* Secondary Modem Codec */
+
+#endif /* __ASM_ARCH_REGS_AC97_H */
diff --git a/drivers/input/touchscreen/mainstone-wm97xx.c b/drivers/input/touchscreen/mainstone-wm97xx.c
index ba648750a8d..1d11e2be9ef 100644
--- a/drivers/input/touchscreen/mainstone-wm97xx.c
+++ b/drivers/input/touchscreen/mainstone-wm97xx.c
@@ -31,7 +31,7 @@
 #include <linux/interrupt.h>
 #include <linux/wm97xx.h>
 #include <linux/io.h>
-#include <mach/pxa-regs.h>
+#include <mach/regs-ac97.h>
 
 #define VERSION		"0.13"
 
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 34c1d94f921..ef6539eea57 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -22,7 +22,7 @@
 
 #include <asm/irq.h>
 #include <mach/hardware.h>
-#include <mach/pxa-regs.h>
+#include <mach/regs-ac97.h>
 #include <mach/pxa2xx-gpio.h>
 #include <mach/audio.h>
 
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index c2635beb4c8..85cf591d4e1 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -22,6 +22,7 @@
 
 #include <mach/hardware.h>
 #include <mach/pxa-regs.h>
+#include <mach/regs-ac97.h>
 #include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a7a3a9c5c6f..5e727393cfd 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -21,6 +21,7 @@
 
 #include <mach/hardware.h>
 #include <mach/pxa-regs.h>
+#include <mach/regs-ac97.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
-- 
cgit v1.2.3-70-g09d2


From 87689d567a45f80416feea0a2aa6d3a2a6b8963a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 2 Dec 2008 16:01:14 +0000
Subject: ASoC: Push platform registration down into the card

As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h                  |  2 +-
 sound/soc/atmel/playpaq_wm8510.c     |  2 +-
 sound/soc/atmel/sam9g20_wm8731.c     |  2 +-
 sound/soc/blackfin/bf5xx-ad1980.c    |  2 +-
 sound/soc/blackfin/bf5xx-ad73311.c   |  2 +-
 sound/soc/blackfin/bf5xx-ssm2602.c   |  2 +-
 sound/soc/davinci/davinci-evm.c      |  2 +-
 sound/soc/davinci/davinci-i2s.c      |  2 +-
 sound/soc/davinci/davinci-sffsdr.c   |  2 +-
 sound/soc/fsl/mpc8610_hpcd.c         |  2 +-
 sound/soc/fsl/soc-of-simple.c        |  2 +-
 sound/soc/omap/n810.c                |  2 +-
 sound/soc/omap/omap2evm.c            |  2 +-
 sound/soc/omap/omap3beagle.c         |  2 +-
 sound/soc/omap/osk5912.c             |  2 +-
 sound/soc/omap/overo.c               |  2 +-
 sound/soc/omap/sdp3430.c             |  2 +-
 sound/soc/pxa/corgi.c                |  2 +-
 sound/soc/pxa/e800_wm9712.c          |  2 +-
 sound/soc/pxa/em-x270.c              |  2 +-
 sound/soc/pxa/palm27x.c              |  2 +-
 sound/soc/pxa/poodle.c               |  2 +-
 sound/soc/pxa/spitz.c                |  2 +-
 sound/soc/pxa/tosa.c                 |  2 +-
 sound/soc/pxa/zylonite.c             |  2 +-
 sound/soc/s3c24xx/ln2440sbc_alc650.c |  2 +-
 sound/soc/s3c24xx/neo1973_wm8753.c   |  2 +-
 sound/soc/s3c24xx/s3c24xx_uda134x.c  |  2 +-
 sound/soc/s3c24xx/smdk2443_wm9710.c  |  2 +-
 sound/soc/sh/sh7760-ac97.c           |  2 +-
 sound/soc/soc-core.c                 | 44 ++++++++++++++++++++----------------
 31 files changed, 55 insertions(+), 49 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 359ec49f8d0..ad8141acd6b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -352,6 +352,7 @@ struct snd_soc_card {
 
 	struct snd_soc_device *socdev;
 
+	struct snd_soc_platform *platform;
 	struct delayed_work delayed_work;
 	struct work_struct deferred_resume_work;
 };
@@ -360,7 +361,6 @@ struct snd_soc_card {
 struct snd_soc_device {
 	struct device *dev;
 	struct snd_soc_card *card;
-	struct snd_soc_platform *platform;
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
 	void *codec_data;
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index d40b5a52a8d..43dd8cee83c 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -363,6 +363,7 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = {
 
 static struct snd_soc_card snd_soc_playpaq = {
 	.name = "LRS_PlayPaq_WM8510",
+	.platform = &at32_soc_platform,
 	.dai_link = &playpaq_wm8510_dai,
 	.num_links = 1,
 };
@@ -378,7 +379,6 @@ static struct wm8510_setup_data playpaq_wm8510_setup = {
 
 static struct snd_soc_device playpaq_wm8510_snd_devdata = {
 	.card = &snd_soc_playpaq,
-	.platform = &at32_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8510,
 	.codec_data = &playpaq_wm8510_setup,
 };
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index fdc1d0206e0..1fb59a9d371 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -244,6 +244,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
 
 static struct snd_soc_card snd_soc_at91sam9g20ek = {
 	.name = "WM8731",
+	.platform = &atmel_soc_platform,
 	.dai_link = &at91sam9g20ek_dai,
 	.num_links = 1,
 };
@@ -255,7 +256,6 @@ static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
 
 static struct snd_soc_device at91sam9g20ek_snd_devdata = {
 	.card = &snd_soc_at91sam9g20ek,
-	.platform = &atmel_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &at91sam9g20ek_wm8731_setup,
 };
diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c
index 36c569a43ce..d8f59127377 100644
--- a/sound/soc/blackfin/bf5xx-ad1980.c
+++ b/sound/soc/blackfin/bf5xx-ad1980.c
@@ -69,13 +69,13 @@ static struct snd_soc_dai_link bf5xx_board_dai = {
 
 static struct snd_soc_card bf5xx_board = {
 	.name = "bf5xx-board",
+	.platform = &bf5xx_ac97_soc_platform,
 	.dai_link = &bf5xx_board_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_board_snd_devdata = {
 	.card = &bf5xx_board,
-	.platform = &bf5xx_ac97_soc_platform,
 	.codec_dev = &soc_codec_dev_ad1980,
 };
 
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 57da1479937..7f2a5e19907 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -192,6 +192,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = {
 
 static struct snd_soc_card bf5xx_ad73311 = {
 	.name = "bf5xx_ad73311",
+	.platform = &bf5xx_i2s_soc_platform,
 	.probe = bf5xx_probe,
 	.dai_link = &bf5xx_ad73311_dai,
 	.num_links = 1,
@@ -199,7 +200,6 @@ static struct snd_soc_card bf5xx_ad73311 = {
 
 static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
 	.card = &bf5xx_ad73311,
-	.platform = &bf5xx_i2s_soc_platform,
 	.codec_dev = &soc_codec_dev_ad73311,
 };
 
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index 0078dfcd95b..bc0cdded711 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -137,13 +137,13 @@ static struct ssm2602_setup_data bf5xx_ssm2602_setup = {
 
 static struct snd_soc_card bf5xx_ssm2602 = {
 	.name = "bf5xx_ssm2602",
+	.platform = &bf5xx_i2s_soc_platform,
 	.dai_link = &bf5xx_ssm2602_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
 	.card = &bf5xx_ssm2602,
-	.platform = &bf5xx_i2s_soc_platform,
 	.codec_dev = &soc_codec_dev_ssm2602,
 	.codec_data = &bf5xx_ssm2602_setup,
 };
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 2ce34d44b15..d87b91179cc 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -130,6 +130,7 @@ static struct snd_soc_dai_link evm_dai = {
 /* davinci-evm audio machine driver */
 static struct snd_soc_card snd_soc_card_evm = {
 	.name = "DaVinci EVM",
+	.platform = &davinci_soc_platform,
 	.dai_link = &evm_dai,
 	.num_links = 1,
 };
@@ -143,7 +144,6 @@ static struct aic3x_setup_data evm_aic3x_setup = {
 /* evm audio subsystem */
 static struct snd_soc_device evm_snd_devdata = {
 	.card = &snd_soc_card_evm,
-	.platform = &davinci_soc_platform,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &evm_aic3x_setup,
 };
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index cf31b3bb96c..8b99efbc64c 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -112,7 +112,7 @@ static void davinci_mcbsp_start(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = socdev->card->platform;
 	u32 w;
 	int ret;
 
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index e95fde1766b..f67579d5276 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -76,6 +76,7 @@ static struct snd_soc_dai_link sffsdr_dai = {
 /* davinci-sffsdr audio machine driver */
 static struct snd_soc_card snd_soc_sffsdr = {
 	.name = "DaVinci SFFSDR",
+	.platform = &davinci_soc_platform,
 	.dai_link = &sffsdr_dai,
 	.num_links = 1,
 };
@@ -91,7 +92,6 @@ static struct pcm3008_setup_data sffsdr_pcm3008_setup = {
 /* sffsdr audio subsystem */
 static struct snd_soc_device sffsdr_snd_devdata = {
 	.card = &snd_soc_sffsdr,
-	.platform = &davinci_soc_platform,
 	.codec_dev = &soc_codec_dev_pcm3008,
 	.codec_data = &sffsdr_pcm3008_setup,
 };
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 1cf4d6eeb53..bcec3f60bad 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -467,7 +467,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 
 	machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
 	machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
-	machine_data->sound_devdata.platform = &fsl_soc_platform;
+	machine_data->machine.platform = &fsl_soc_platform;
 
 	sound_device->dev.platform_data = machine_data;
 
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c
index 53be6491320..8bc5cd9e972 100644
--- a/sound/soc/fsl/soc-of-simple.c
+++ b/sound/soc/fsl/soc-of-simple.c
@@ -158,7 +158,7 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform,
 
 	of_soc->platform_node = node;
 	of_soc->dai_link.cpu_dai = cpu_dai;
-	of_soc->device.platform = platform;
+	of_soc->card.platform = platform;
 	of_soc->card.name = of_soc->dai_link.cpu_dai->name;
 
 	/* Now try to register the SoC device */
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 18e2062e3a1..25593fee912 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -288,6 +288,7 @@ static struct snd_soc_dai_link n810_dai = {
 /* Audio machine driver */
 static struct snd_soc_card snd_soc_n810 = {
 	.name = "N810",
+	.platform = &omap_soc_platform,
 	.dai_link = &n810_dai,
 	.num_links = 1,
 };
@@ -303,7 +304,6 @@ static struct aic3x_setup_data n810_aic33_setup = {
 /* Audio subsystem */
 static struct snd_soc_device n810_snd_devdata = {
 	.card = &snd_soc_n810,
-	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &n810_aic33_setup,
 };
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index 7b160f9d83f..0c2322dcf02 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -93,6 +93,7 @@ static struct snd_soc_dai_link omap2evm_dai = {
 /* Audio machine driver */
 static struct snd_soc_card snd_soc_omap2evm = {
 	.name = "omap2evm",
+	.platform = &omap_soc_platform,
 	.dai_link = &omap2evm_dai,
 	.num_links = 1,
 };
@@ -100,7 +101,6 @@ static struct snd_soc_card snd_soc_omap2evm = {
 /* Audio subsystem */
 static struct snd_soc_device omap2evm_snd_devdata = {
 	.card = &snd_soc_omap2evm,
-	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
 
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index 3ed25464627..fd24a4acd2f 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -90,6 +90,7 @@ static struct snd_soc_dai_link omap3beagle_dai = {
 /* Audio machine driver */
 static struct snd_soc_card snd_soc_omap3beagle = {
 	.name = "omap3beagle",
+	.platform = &omap_soc_platform,
 	.dai_link = &omap3beagle_dai,
 	.num_links = 1,
 };
@@ -97,7 +98,6 @@ static struct snd_soc_card snd_soc_omap3beagle = {
 /* Audio subsystem */
 static struct snd_soc_device omap3beagle_snd_devdata = {
 	.card = &snd_soc_omap3beagle,
-	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
 
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 7a8f14d0c77..845bf41335b 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -145,6 +145,7 @@ static struct snd_soc_dai_link osk_dai = {
 /* Audio machine driver */
 static struct snd_soc_card snd_soc_card_osk = {
 	.name = "OSK5912",
+	.platform = &omap_soc_platform,
 	.dai_link = &osk_dai,
 	.num_links = 1,
 };
@@ -152,7 +153,6 @@ static struct snd_soc_card snd_soc_card_osk = {
 /* Audio subsystem */
 static struct snd_soc_device osk_snd_devdata = {
 	.card = &snd_soc_card_osk,
-	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_tlv320aic23,
 };
 
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index eea0c372bb3..a72dc4e159e 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -90,6 +90,7 @@ static struct snd_soc_dai_link overo_dai = {
 /* Audio machine driver */
 static struct snd_soc_card snd_soc_card_overo = {
 	.name = "overo",
+	.platform = &omap_soc_platform,
 	.dai_link = &overo_dai,
 	.num_links = 1,
 };
@@ -97,7 +98,6 @@ static struct snd_soc_card snd_soc_card_overo = {
 /* Audio subsystem */
 static struct snd_soc_device overo_snd_devdata = {
 	.card = &snd_soc_card_overo,
-	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
 
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 85fd160bca1..ad97836818b 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -93,6 +93,7 @@ static struct snd_soc_dai_link sdp3430_dai = {
 /* Audio machine driver */
 static struct snd_soc_machine snd_soc_machine_sdp3430 = {
 	.name = "SDP3430",
+	.platform = &omap_soc_platform,
 	.dai_link = &sdp3430_dai,
 	.num_links = 1,
 };
@@ -100,7 +101,6 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = {
 /* Audio subsystem */
 static struct snd_soc_device sdp3430_snd_devdata = {
 	.machine = &snd_soc_machine_sdp3430,
-	.platform = &omap_soc_platform,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
 
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index e56bf4b6c2a..1ba25a55952 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -312,6 +312,7 @@ static struct snd_soc_dai_link corgi_dai = {
 /* corgi audio machine driver */
 static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &corgi_dai,
 	.num_links = 1,
 };
@@ -325,7 +326,6 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
 	.card = &snd_soc_corgi,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,
 };
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 60c64861512..2e3386dfa0f 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -42,13 +42,13 @@ static struct snd_soc_dai_link e800_dai[] = {
 
 static struct snd_soc_card e800 = {
 	.name = "Toshiba e800",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = e800_dai,
 	.num_links = ARRAY_SIZE(e800_dai),
 };
 
 static struct snd_soc_device e800_snd_devdata = {
 	.card = &e800,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index d6884b755d5..fe4a729ea64 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -54,13 +54,13 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 
 static struct snd_soc_card em_x270 = {
 	.name = "EM-X270",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = em_x270_dai,
 	.num_links = ARRAY_SIZE(em_x270_dai),
 };
 
 static struct snd_soc_device em_x270_snd_devdata = {
 	.card = &em_x270,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 3bb8879ac8a..4a9cf3083af 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -191,13 +191,13 @@ static struct snd_soc_dai_link palm27x_dai[] = {
 
 static struct snd_soc_card palm27x_asoc = {
 	.name = "Palm/PXA27x",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = palm27x_dai,
 	.num_links = ARRAY_SIZE(palm27x_dai),
 };
 
 static struct snd_soc_device palm27x_snd_devdata = {
 	.card = &palm27x_asoc,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 03b510ab282..6e9827189ff 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -278,6 +278,7 @@ static struct snd_soc_dai_link poodle_dai = {
 /* poodle audio machine driver */
 static struct snd_soc_card snd_soc_poodle = {
 	.name = "Poodle",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &poodle_dai,
 	.num_links = 1,
 };
@@ -291,7 +292,6 @@ static struct wm8731_setup_data poodle_wm8731_setup = {
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
 	.card = &snd_soc_poodle,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &poodle_wm8731_setup,
 };
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 579d93368f1..a3b9e6bdf97 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -321,6 +321,7 @@ static struct snd_soc_dai_link spitz_dai = {
 /* spitz audio machine driver */
 static struct snd_soc_card snd_soc_spitz = {
 	.name = "Spitz",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &spitz_dai,
 	.num_links = 1,
 };
@@ -334,7 +335,6 @@ static struct wm8750_setup_data spitz_wm8750_setup = {
 /* spitz audio subsystem */
 static struct snd_soc_device spitz_snd_devdata = {
 	.card = &snd_soc_spitz,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8750,
 	.codec_data = &spitz_wm8750_setup,
 };
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 48242b32a28..c77194f74c9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -252,6 +252,7 @@ static int tosa_remove(struct platform_device *dev)
 
 static struct snd_soc_card tosa = {
 	.name = "Tosa",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
 	.probe = tosa_probe,
@@ -260,7 +261,6 @@ static struct snd_soc_card tosa = {
 
 static struct snd_soc_device tosa_snd_devdata = {
 	.card = &tosa,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 842d6500d61..f8e9ecd589d 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -175,13 +175,13 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 
 static struct snd_soc_card zylonite = {
 	.name = "Zylonite",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = zylonite_dai,
 	.num_links = ARRAY_SIZE(zylonite_dai),
 };
 
 static struct snd_soc_device zylonite_snd_ac97_devdata = {
 	.card = &zylonite,
-	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm9713,
 };
 
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
index a70cbc0fa07..12c71482d25 100644
--- a/sound/soc/s3c24xx/ln2440sbc_alc650.c
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -40,13 +40,13 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = {
 
 static struct snd_soc_card ln2440sbc = {
 	.name = "LN2440SBC",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = ln2440sbc_dai,
 	.num_links = ARRAY_SIZE(ln2440sbc_dai),
 };
 
 static struct snd_soc_device ln2440sbc_snd_ac97_devdata = {
 	.card = &ln2440sbc,
-	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 3df2224a672..45bb12e8ea4 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -580,6 +580,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 
 static struct snd_soc_card neo1973 = {
 	.name = "neo1973",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = neo1973_dai,
 	.num_links = ARRAY_SIZE(neo1973_dai),
 };
@@ -591,7 +592,6 @@ static struct wm8753_setup_data neo1973_wm8753_setup = {
 
 static struct snd_soc_device neo1973_snd_devdata = {
 	.card = &neo1973,
-	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8753,
 	.codec_data = &neo1973_wm8753_setup,
 };
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index 23325fca1f6..a0a4d1832a1 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -234,6 +234,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
 
 static struct snd_soc_card snd_soc_s3c24xx_uda134x = {
 	.name = "S3C24XX_UDA134X",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = &s3c24xx_uda134x_dai_link,
 	.num_links = 1,
 };
@@ -271,7 +272,6 @@ static struct uda134x_platform_data s3c24xx_uda134x = {
 
 static struct snd_soc_device s3c24xx_uda134x_snd_devdata = {
 	.card = &snd_soc_s3c24xx_uda134x,
-	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_uda134x,
 	.codec_data = &s3c24xx_uda134x,
 };
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index 3d2e6a0417e..a2a4f5323c1 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -36,13 +36,13 @@ static struct snd_soc_dai_link smdk2443_dai[] = {
 
 static struct snd_soc_card smdk2443 = {
 	.name = "SMDK2443",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = smdk2443_dai,
 	.num_links = ARRAY_SIZE(smdk2443_dai),
 };
 
 static struct snd_soc_device smdk2443_snd_ac97_devdata = {
 	.card = &smdk2443,
-	.platform = &s3c24xx_soc_platform,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 8b44f9c8a9f..ce7f95b59de 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -40,13 +40,13 @@ static struct snd_soc_dai_link sh7760_ac97_dai = {
 
 static struct snd_soc_card sh7760_ac97_soc_machine  = {
 	.name = "SH7760 AC97",
+	.platform = &sh7760_soc_platform,
 	.dai_link = &sh7760_ac97_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device sh7760_ac97_snd_devdata = {
 	.card = &sh7760_ac97_soc_machine,
-	.platform = &sh7760_soc_platform,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c4b22e6984e..fe89260c902 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -109,9 +109,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card = socdev->card;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
@@ -302,7 +303,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -370,7 +371,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -464,7 +465,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
@@ -534,7 +536,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -568,8 +571,9 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card= socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
@@ -610,7 +614,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
 	int i;
@@ -686,7 +690,7 @@ static void soc_resume_deferred(struct work_struct *work)
 						 struct snd_soc_card,
 						 deferred_resume_work);
 	struct snd_soc_device *socdev = card->socdev;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct platform_device *pdev = to_platform_device(socdev->dev);
@@ -770,7 +774,7 @@ static int soc_probe(struct platform_device *pdev)
 	int ret = 0, i;
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
 	/* Bodge while we push things out of socdev */
@@ -835,7 +839,7 @@ static int soc_remove(struct platform_device *pdev)
 	int i;
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
 	run_delayed_work(&card->delayed_work);
@@ -875,6 +879,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	struct snd_soc_dai_link *dai_link, int num)
 {
 	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
 	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
 	struct snd_soc_pcm_runtime *rtd;
@@ -910,13 +916,13 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 
 	dai_link->pcm = pcm;
 	pcm->private_data = rtd;
-	soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
-	soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
-	soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
-	soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
-	soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
-	soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
-	soc_pcm_ops.page = socdev->platform->pcm_ops->page;
+	soc_pcm_ops.mmap = platform->pcm_ops->mmap;
+	soc_pcm_ops.pointer = platform->pcm_ops->pointer;
+	soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
+	soc_pcm_ops.copy = platform->pcm_ops->copy;
+	soc_pcm_ops.silence = platform->pcm_ops->silence;
+	soc_pcm_ops.ack = platform->pcm_ops->ack;
+	soc_pcm_ops.page = platform->pcm_ops->page;
 
 	if (playback)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
@@ -924,14 +930,14 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	if (capture)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
 
-	ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
+	ret = platform->pcm_new(codec->card, codec_dai, pcm);
 	if (ret < 0) {
 		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
 		kfree(rtd);
 		return ret;
 	}
 
-	pcm->private_free = socdev->platform->pcm_free;
+	pcm->private_free = platform->pcm_free;
 	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
 		cpu_dai->name);
 	return ret;
-- 
cgit v1.2.3-70-g09d2


From dc7d7b830ee1f4111696e73d1c25da683b461548 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 18:21:52 +0000
Subject: ASoC: Remove platform device from DAI suspend and resume operations

None of the DAIs use it except s3c2412-i2s which only uses it for
dev_() printouts.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h         |  6 ++----
 sound/soc/atmel/atmel_ssc_dai.c |  6 ++----
 sound/soc/au1x/psc-ac97.c       |  6 ++----
 sound/soc/au1x/psc-i2s.c        |  6 ++----
 sound/soc/blackfin/bf5xx-ac97.c |  6 ++----
 sound/soc/blackfin/bf5xx-i2s.c  |  6 ++----
 sound/soc/pxa/pxa-ssp.c         |  6 ++----
 sound/soc/pxa/pxa2xx-ac97.c     |  6 ++----
 sound/soc/pxa/pxa2xx-i2s.c      |  6 ++----
 sound/soc/s3c24xx/s3c2412-i2s.c | 16 +++++++---------
 sound/soc/s3c24xx/s3c24xx-i2s.c |  6 ++----
 sound/soc/soc-core.c            |  8 ++++----
 12 files changed, 31 insertions(+), 53 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index a01a24b1019..e2d5f76838c 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -191,10 +191,8 @@ struct snd_soc_dai {
 		     struct snd_soc_dai *dai);
 	void (*remove)(struct platform_device *pdev,
 		       struct snd_soc_dai *dai);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
 
 	/* ops */
 	struct snd_soc_dai_ops ops;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 0bb18dfa949..d9b874c5bf3 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -628,8 +628,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 
 
 #ifdef CONFIG_PM
-static int atmel_ssc_suspend(struct platform_device *pdev,
-			    struct snd_soc_dai *cpu_dai)
+static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
 {
 	struct atmel_ssc_info *ssc_p;
 
@@ -657,8 +656,7 @@ static int atmel_ssc_suspend(struct platform_device *pdev,
 
 
 
-static int atmel_ssc_resume(struct platform_device *pdev,
-			   struct snd_soc_dai *cpu_dai)
+static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
 {
 	struct atmel_ssc_info *ssc_p;
 	u32 cr;
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a0bcfeaf5f8..a1e824d29cf 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -314,8 +314,7 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev,
 	au1xpsc_ac97_workdata = NULL;
 }
 
-static int au1xpsc_ac97_suspend(struct platform_device *pdev,
-				struct snd_soc_dai *dai)
+static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
 {
 	/* save interesting registers and disable PSC */
 	au1xpsc_ac97_workdata->pm[0] =
@@ -329,8 +328,7 @@ static int au1xpsc_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_ac97_resume(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 {
 	/* restore PSC clock config */
 	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index f4217e70a78..16f97462ea1 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -339,8 +339,7 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev,
 	au1xpsc_i2s_workdata = NULL;
 }
 
-static int au1xpsc_i2s_suspend(struct platform_device *pdev,
-			       struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	/* save interesting register and disable PSC */
 	au1xpsc_i2s_workdata->pm[0] =
@@ -354,8 +353,7 @@ static int au1xpsc_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	/* select I2S mode and PSC clock */
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 709bdf08e39..c602ce109d5 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -269,8 +269,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 #ifdef CONFIG_PM
-static int bf5xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
@@ -285,8 +284,7 @@ static int bf5xx_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int bf5xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	int ret;
 	struct sport_device *sport =
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 6e5036bf924..9f8ce87cc6c 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -222,16 +222,14 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void bf5xx_i2s_remove(struct platform_device *pdev,
-			   struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	peripheral_free_list(&sport_req[sport_num][0]);
 }
 
 #ifdef CONFIG_PM
-static int bf5xx_i2s_suspend(struct platform_device *dev,
-			     struct snd_soc_dai *dai)
+static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 402fc5ba65e..73fa10defcc 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -244,8 +244,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 
 #ifdef CONFIG_PM
 
-static int pxa_ssp_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *cpu_dai)
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
 {
 	struct ssp_priv *priv = cpu_dai->private_data;
 
@@ -257,8 +256,7 @@ static int pxa_ssp_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int pxa_ssp_resume(struct platform_device *pdev,
-	struct snd_soc_dai *cpu_dai)
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
 {
 	struct ssp_priv *priv = cpu_dai->private_data;
 
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index bffbe288634..8eed80d5675 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -87,14 +87,12 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
 };
 
 #ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_suspend();
 }
 
-static int pxa2xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_resume();
 }
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index f9a9e2ebafa..314973ace6d 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -293,8 +293,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
 }
 
 #ifdef CONFIG_PM
-static int pxa2xx_i2s_suspend(struct platform_device *dev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -311,8 +310,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
 	return 0;
 }
 
-static int pxa2xx_i2s_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 1c741047ae3..75f87c3c74d 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -649,8 +649,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct platform_device *dev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 	u32 iismod;
@@ -665,25 +664,24 @@ static int s3c2412_i2s_suspend(struct platform_device *dev,
 		iismod = readl(i2s->regs + S3C2412_IISMOD);
 
 		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__);
+			pr_warning("%s: RXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__);
+			pr_warning("%s: TXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			dev_warn(&dev->dev, "%s: IIS active\n", __func__);
+			pr_warning("%s: IIS active\n", __func__);
 	}
 
 	return 0;
 }
 
-static int s3c2412_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 
-	dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n",
-		 dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
 
 	if (dai->active) {
 		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 8d9135f41bc..45fe8f7c88a 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -419,8 +419,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c24xx_i2s_suspend(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 
@@ -434,8 +433,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int s3c24xx_i2s_resume(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 	clk_enable(s3c24xx_i2s.iis_clk);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f83852f1146..2f2a8d93bbf 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -650,7 +650,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->suspend && !cpu_dai->ac97_control)
-			cpu_dai->suspend(pdev, cpu_dai);
+			cpu_dai->suspend(cpu_dai);
 		if (platform->suspend)
 			platform->suspend(cpu_dai);
 	}
@@ -676,7 +676,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->suspend && cpu_dai->ac97_control)
-			cpu_dai->suspend(pdev, cpu_dai);
+			cpu_dai->suspend(cpu_dai);
 	}
 
 	if (card->suspend_post)
@@ -712,7 +712,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->resume && cpu_dai->ac97_control)
-			cpu_dai->resume(pdev, cpu_dai);
+			cpu_dai->resume(cpu_dai);
 	}
 
 	if (codec_dev->resume)
@@ -739,7 +739,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->resume && !cpu_dai->ac97_control)
-			cpu_dai->resume(pdev, cpu_dai);
+			cpu_dai->resume(cpu_dai);
 		if (platform->resume)
 			platform->resume(cpu_dai);
 	}
-- 
cgit v1.2.3-70-g09d2


From 3f4b783cfdebb559814690572041a17bc9744cf3 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 19:26:35 +0000
Subject: ASoC: Register platform DAIs

Register all platform DAIs with the core.  In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel_ssc_dai.c  | 11 +++++++++++
 sound/soc/au1x/psc-ac97.c        |  3 ++-
 sound/soc/au1x/psc-i2s.c         |  3 ++-
 sound/soc/blackfin/bf5xx-ac97.c  | 12 ++++++++++++
 sound/soc/blackfin/bf5xx-i2s.c   | 12 ++++++++++++
 sound/soc/davinci/davinci-i2s.c  | 12 ++++++++++++
 sound/soc/fsl/fsl_ssi.c          | 10 ++++++++++
 sound/soc/omap/omap-mcbsp.c      | 13 +++++++++++++
 sound/soc/pxa/pxa-ssp.c          | 12 ++++++++++++
 sound/soc/pxa/pxa2xx-ac97.c      | 12 ++++++++++++
 sound/soc/pxa/pxa2xx-i2s.c       | 13 ++++++++++++-
 sound/soc/s3c24xx/s3c2412-i2s.c  | 13 +++++++++++++
 sound/soc/s3c24xx/s3c2443-ac97.c | 13 +++++++++++++
 sound/soc/s3c24xx/s3c24xx-i2s.c  | 12 ++++++++++++
 sound/soc/sh/hac.c               | 12 ++++++++++++
 sound/soc/sh/ssi.c               | 12 ++++++++++++
 16 files changed, 172 insertions(+), 3 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index d9b874c5bf3..c9f02edd730 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -772,6 +772,17 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 };
 EXPORT_SYMBOL_GPL(atmel_ssc_dai);
 
+static int __devinit atmel_ssc_modinit(void)
+{
+	return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_init(snd_soc_init);
+
+static void __exit snd_soc_exit(void)
+{
+	snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+
 /* Module information */
 MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com");
 MODULE_DESCRIPTION("ATMEL SSC ASoC Interface");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a1e824d29cf..f0e30aec7f2 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -371,11 +371,12 @@ EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 static int __init au1xpsc_ac97_init(void)
 {
 	au1xpsc_ac97_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_ac97_dai);
 }
 
 static void __exit au1xpsc_ac97_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
 }
 
 module_init(au1xpsc_ac97_init);
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 16f97462ea1..f916de4400e 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -396,11 +396,12 @@ EXPORT_SYMBOL(au1xpsc_i2s_dai);
 static int __init au1xpsc_i2s_init(void)
 {
 	au1xpsc_i2s_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_i2s_dai);
 }
 
 static void __exit au1xpsc_i2s_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
 }
 
 module_init(au1xpsc_i2s_init);
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index c602ce109d5..ad3efeeb6d4 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -431,6 +431,18 @@ struct snd_soc_dai bfin_ac97_dai = {
 };
 EXPORT_SYMBOL_GPL(bfin_ac97_dai);
 
+static int __devinit bfin_ac97_init(void)
+{
+	return snd_soc_register_dai(&bfin_ac97_dai);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_dai(&bfin_ac97_dai);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Roy Huang");
 MODULE_DESCRIPTION("AC97 driver for ADI Blackfin");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 9f8ce87cc6c..0d58d2b6db6 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -313,6 +313,18 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
+static int __devinit bfin_i2s_init(void)
+{
+	return snd_soc_register_dai(&bfin_i2s_dai);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&bfin_i2s_dai);
+}
+module_exit(bfin_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("I2S driver for ADI Blackfin");
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 8b99efbc64c..d89fc2f009a 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -481,6 +481,18 @@ struct snd_soc_dai davinci_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
+static int __devinit davinci_i2s_init(void)
+{
+	return snd_soc_register_dai(&davinci_i2s_dai);
+}
+module_init(davinci_i2s_init);
+
+static void __exit davinci_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&davinci_i2s_dai);
+}
+module_exit(davinci_i2s_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 52c290bb47b..c6d6eb71dc1 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -673,6 +673,14 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	fsl_ssi_dai->private_data = ssi_private;
 	fsl_ssi_dai->name = ssi_private->name;
 	fsl_ssi_dai->id = ssi_info->id;
+	fsl_ssi_dai->dev = ssi_info->dev;
+
+	ret = snd_soc_register_dai(fsl_ssi_dai);
+	if (ret != 0) {
+		dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret);
+		kfree(fsl_ssi_dai);
+		return NULL;
+	}
 
 	return fsl_ssi_dai;
 }
@@ -690,6 +698,8 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 
 	device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
 
+	snd_soc_unregister_dai(&ssi_private->cpu_dai);
+
 	kfree(ssi_private);
 }
 EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index e8f1314762d..41cab203416 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -499,6 +499,19 @@ struct snd_soc_dai omap_mcbsp_dai[] = {
 
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
+static int __devinit omap_mcbsp_init(void)
+{
+	return snd_soc_register_dais(omap_mcbsp_dai,
+				     ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_init(omap_mcbsp_init);
+
+static void __exit omap_mcbsp_exit(void)
+{
+	snd_soc_unregister_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_exit(omap_mcbsp_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 73fa10defcc..3587f2fae5f 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -913,6 +913,18 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 };
 EXPORT_SYMBOL_GPL(pxa_ssp_dai);
 
+static int __devinit pxa_ssp_init(void)
+{
+	return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_init(pxa_ssp_init);
+
+static void __exit pxa_ssp_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_exit(pxa_ssp_exit);
+
 /* Module information */
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 8eed80d5675..f6249d5b492 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -228,6 +228,18 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __devinit pxa_ac97_init(void)
+{
+	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_init(pxa_ac97_init);
+
+static void __exit pxa_ac97_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_exit(pxa_ac97_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 314973ace6d..517991fb109 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -364,12 +364,23 @@ EXPORT_SYMBOL_GPL(pxa_i2s_dai);
 
 static int pxa2xx_i2s_probe(struct platform_device *dev)
 {
+	int ret;
+
 	clk_i2s = clk_get(&dev->dev, "I2SCLK");
-	return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
+
+	pxa_i2s_dai.dev = &dev->dev;
+	ret = snd_soc_register_dai(&pxa_i2s_dai);
+	if (ret != 0)
+		clk_put(clk_i2s);
+
+	return ret;
 }
 
 static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
 {
+	snd_soc_unregister_dai(&pxa_i2s_dai);
 	clk_put(clk_i2s);
 	clk_i2s = ERR_PTR(-ENOENT);
 	return 0;
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 75f87c3c74d..2cf05079156 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -736,6 +736,19 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
+static int __devinit s3c2412_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c2412_i2s_dai);
+}
+module_init(s3c2412_i2s_init);
+
+static void __exit s3c2412_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c2412_i2s_dai);
+}
+module_exit(s3c2412_i2s_exit);
+
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index f0bc9b7e084..7360c6d913b 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -396,6 +396,19 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __devinit s3c2443_ac97_init(void)
+{
+	return snd_soc_register_dai(&s3c2443_ac97_dai);
+}
+module_init(s3c2443_ac97_init);
+
+static void __exit s3c2443_ac97_exit(void)
+{
+	snd_soc_unregister_dai(&s3c2443_ac97_dai);
+}
+module_exit(s3c2443_ac97_exit);
+
+
 MODULE_AUTHOR("Graeme Gregory");
 MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 45fe8f7c88a..897b1ac92ce 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -482,6 +482,18 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
+static int __devinit s3c24xx_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c24xx_i2s_dai);
+}
+module_init(s3c24xx_i2s_init);
+
+static void __exit s3c24xx_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c24xx_i2s_dai);
+}
+module_exit(s3c24xx_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index c435913c60e..9169bad1acf 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -314,6 +314,18 @@ struct snd_soc_dai sh4_hac_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_hac_dai);
 
+static int __devinit sh4_hac_init(void)
+{
+	return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_init(sh4_hac_init);
+
+static void __exit sh4_hac_exit(void)
+{
+	snd_soc_unregister_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_exit(sh4_hac_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index fed544a3def..9093588d4d0 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -392,6 +392,18 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_ssi_dai);
 
+static int __devinit sh4_ssi_init(void)
+{
+	return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_init(sh4_ssi_init);
+
+static void __exit sh4_ssi_exit(void)
+{
+	snd_soc_unregister_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_exit(sh4_ssi_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
-- 
cgit v1.2.3-70-g09d2


From 958e792c7c8f06a9e666adb0ed94fff2cf90156f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 19:58:17 +0000
Subject: ASoC: Register platform drivers

This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel-pcm.c         | 12 ++++++++++++
 sound/soc/au1x/dbdma2.c             |  3 ++-
 sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 ++++++++++++
 sound/soc/blackfin/bf5xx-i2s-pcm.c  | 12 ++++++++++++
 sound/soc/davinci/davinci-pcm.c     | 12 ++++++++++++
 sound/soc/fsl/fsl_dma.c             | 12 ++++++++++++
 sound/soc/fsl/mpc5200_psc_i2s.c     |  4 ++++
 sound/soc/omap/omap-pcm.c           | 12 ++++++++++++
 sound/soc/pxa/pxa2xx-pcm.c          | 12 ++++++++++++
 sound/soc/s3c24xx/s3c24xx-pcm.c     | 12 ++++++++++++
 sound/soc/sh/dma-sh7760.c           | 12 ++++++++++++
 11 files changed, 114 insertions(+), 1 deletion(-)

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 8507aa1cd81..d6bcb4e6fda 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -477,6 +477,18 @@ struct snd_soc_platform atmel_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(atmel_soc_platform);
 
+static int __devinit atmel_pcm_modinit(void)
+{
+	return snd_soc_register_platform(&atmel_soc_platform);
+}
+module_init(atmel_pcm_modinit);
+
+static void __exit atmel_pcm_exit(void)
+{
+	snd_soc_unregister_platform(&atmel_soc_platform);
+}
+module_exit(atmel_pcm_modexit);
+
 MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
 MODULE_DESCRIPTION("Atmel PCM module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 1466d932880..74c823d60f9 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -406,11 +406,12 @@ static int __init au1xpsc_audio_dbdma_init(void)
 {
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
-	return 0;
+	return snd_soc_register_platform(&au1xpsc_soc_platform);
 }
 
 static void __exit au1xpsc_audio_dbdma_exit(void)
 {
+	snd_soc_unregister_platform(&au1xpsc_soc_platform);
 }
 
 module_init(au1xpsc_audio_dbdma_init);
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index d3d51bcb456..5b27e0d9d0e 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -451,6 +451,18 @@ struct snd_soc_platform bf5xx_ac97_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform);
 
+static int __devinit bfin_ac97_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_ac97_soc_platform);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_ac97_soc_platform);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 61fccf92519..c58b12a4487 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -283,6 +283,18 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
 
+static int __devinit bfin_i2s_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_i2s_soc_platform);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_i2s_soc_platform);
+}
+module_exit(bfin_i2s_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 76feaa65737..f1b6e02d24e 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -384,6 +384,18 @@ struct snd_soc_platform davinci_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(davinci_soc_platform);
 
+static int __devinit davinci_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&davinci_soc_platform);
+}
+module_init(davinci_soc_platform_init);
+
+static void __exit davinci_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&davinci_soc_platform);
+}
+module_exit(davinci_soc_platform_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index bf92331b476..646c807163a 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -853,6 +853,18 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info)
 }
 EXPORT_SYMBOL_GPL(fsl_dma_configure);
 
+static int __devinit fsl_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&fsl_soc_platform);
+}
+module_init(fsl_soc_platform_init);
+
+static void __exit fsl_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&fsl_soc_platform);
+}
+module_exit(fsl_soc_platform_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9ad8f9a2d8e..9eb1ce185bd 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -828,6 +828,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 	if (rc)
 		dev_info(psc_i2s->dev, "error creating sysfs files\n");
 
+	snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
+
 	/* Tell the ASoC OF helpers about it */
 	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
 				     &psc_i2s->dai);
@@ -841,6 +843,8 @@ static int __devexit psc_i2s_of_remove(struct of_device *op)
 
 	dev_dbg(&op->dev, "psc_i2s_remove()\n");
 
+	snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
+
 	bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
 	bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e9084fdd208..9940de29631 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -354,6 +354,18 @@ struct snd_soc_platform omap_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(omap_soc_platform);
 
+static int __devinit omap_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&omap_soc_platform);
+}
+module_init(omap_soc_platform_init);
+
+static void __exit omap_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&omap_soc_platform);
+}
+module_exit(omap_soc_platform_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 0f6b7bb2d44..4fa1578f5d4 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -118,6 +118,18 @@ struct snd_soc_platform pxa2xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
 
+static int __devinit pxa2xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&pxa2xx_soc_platform);
+}
+module_init(pxa2xx_soc_platform_init);
+
+static void __exit pxa2xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&pxa2xx_soc_platform);
+}
+module_exit(pxa2xx_soc_platform_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index e13e614bada..ea5a9caec13 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -465,6 +465,18 @@ struct snd_soc_platform s3c24xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
 
+static int __devinit s3c24xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&s3c24xx_soc_platform);
+}
+module_init(s3c24xx_soc_platform_init);
+
+static void __exit s3c24xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&s3c24xx_soc_platform);
+}
+module_exit(s3c24xx_soc_platform_exit);
+
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 9faa12622d0..39ffca0933a 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -348,6 +348,18 @@ struct snd_soc_platform sh7760_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(sh7760_soc_platform);
 
+static int __devinit sh7760_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&sh7760_soc_platform);
+}
+module_init(sh7760_soc_platform_init);
+
+static void __exit sh7760_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&sh7760_soc_platform);
+}
+module_exit(sh7760_soc_platform_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
-- 
cgit v1.2.3-70-g09d2


From c9b3a40ff2b3dea9914e36965a17c802650bb603 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 10 Dec 2008 07:47:22 +0100
Subject: ALSA: ASoC - Fix wrong section types

The module init entries should be __init instead of __devinit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/atmel/atmel-pcm.c         | 2 +-
 sound/soc/atmel/atmel_ssc_dai.c     | 2 +-
 sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +-
 sound/soc/blackfin/bf5xx-ac97.c     | 2 +-
 sound/soc/blackfin/bf5xx-i2s-pcm.c  | 2 +-
 sound/soc/blackfin/bf5xx-i2s.c      | 2 +-
 sound/soc/codecs/ad73311.c          | 2 +-
 sound/soc/codecs/ak4535.c           | 2 +-
 sound/soc/codecs/cs4270.c           | 2 +-
 sound/soc/codecs/pcm3008.c          | 2 +-
 sound/soc/codecs/ssm2602.c          | 2 +-
 sound/soc/codecs/tlv320aic23.c      | 2 +-
 sound/soc/codecs/tlv320aic3x.c      | 2 +-
 sound/soc/codecs/uda134x.c          | 2 +-
 sound/soc/codecs/uda1380.c          | 2 +-
 sound/soc/codecs/wm8510.c           | 2 +-
 sound/soc/codecs/wm8580.c           | 2 +-
 sound/soc/codecs/wm8728.c           | 2 +-
 sound/soc/codecs/wm8731.c           | 2 +-
 sound/soc/codecs/wm8750.c           | 2 +-
 sound/soc/codecs/wm8753.c           | 2 +-
 sound/soc/codecs/wm8900.c           | 2 +-
 sound/soc/codecs/wm8903.c           | 2 +-
 sound/soc/codecs/wm8971.c           | 2 +-
 sound/soc/codecs/wm8990.c           | 2 +-
 sound/soc/davinci/davinci-i2s.c     | 2 +-
 sound/soc/davinci/davinci-pcm.c     | 2 +-
 sound/soc/fsl/fsl_dma.c             | 2 +-
 sound/soc/omap/omap-mcbsp.c         | 2 +-
 sound/soc/omap/omap-pcm.c           | 2 +-
 sound/soc/pxa/pxa-ssp.c             | 2 +-
 sound/soc/pxa/pxa2xx-ac97.c         | 2 +-
 sound/soc/pxa/pxa2xx-pcm.c          | 2 +-
 sound/soc/s3c24xx/s3c2412-i2s.c     | 2 +-
 sound/soc/s3c24xx/s3c2443-ac97.c    | 2 +-
 sound/soc/s3c24xx/s3c24xx-i2s.c     | 2 +-
 sound/soc/s3c24xx/s3c24xx-pcm.c     | 2 +-
 sound/soc/sh/dma-sh7760.c           | 2 +-
 sound/soc/sh/hac.c                  | 2 +-
 sound/soc/sh/ssi.c                  | 2 +-
 sound/soc/soc-core.c                | 2 +-
 41 files changed, 41 insertions(+), 41 deletions(-)

(limited to 'sound/soc/pxa')

diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 027eb13f9dd..1fac5efd285 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -477,7 +477,7 @@ struct snd_soc_platform atmel_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(atmel_soc_platform);
 
-static int __devinit atmel_pcm_modinit(void)
+static int __init atmel_pcm_modinit(void)
 {
 	return snd_soc_register_platform(&atmel_soc_platform);
 }
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 87904b6ab8c..c5d67900d66 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -772,7 +772,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 };
 EXPORT_SYMBOL_GPL(atmel_ssc_dai);
 
-static int __devinit atmel_ssc_modinit(void)
+static int __init atmel_ssc_modinit(void)
 {
 	return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
 }
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 5b27e0d9d0e..8067cfafa3a 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -451,7 +451,7 @@ struct snd_soc_platform bf5xx_ac97_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform);
 
-static int __devinit bfin_ac97_init(void)
+static int __init bfin_ac97_init(void)
 {
 	return snd_soc_register_platform(&bf5xx_ac97_soc_platform);
 }
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index ad3efeeb6d4..3be2be60576 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -431,7 +431,7 @@ struct snd_soc_dai bfin_ac97_dai = {
 };
 EXPORT_SYMBOL_GPL(bfin_ac97_dai);
 
-static int __devinit bfin_ac97_init(void)
+static int __init bfin_ac97_init(void)
 {
 	return snd_soc_register_dai(&bfin_ac97_dai);
 }
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index c58b12a4487..53d290b3ea4 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -283,7 +283,7 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
 
-static int __devinit bfin_i2s_init(void)
+static int __init bfin_i2s_init(void)
 {
 	return snd_soc_register_platform(&bf5xx_i2s_soc_platform);
 }
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 0d58d2b6db6..c17b131f662 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -313,7 +313,7 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
-static int __devinit bfin_i2s_init(void)
+static int __init bfin_i2s_init(void)
 {
 	return snd_soc_register_dai(&bfin_i2s_dai);
 }
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index e32f55034e6..b09289a1e55 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -98,7 +98,7 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
 
-static int __devinit ad73311_init(void)
+static int __init ad73311_init(void)
 {
 	return snd_soc_register_dai(&ad73311_dai);
 }
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 94148fba911..81300d8d42c 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -688,7 +688,7 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
 
-static int __devinit ak4535_modinit(void)
+static int __init ak4535_modinit(void)
 {
 	return snd_soc_register_dai(&ak4535_dai);
 }
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 73aaf249d78..f1aa0c34421 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -774,7 +774,7 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
-static int __devinit cs4270_init(void)
+static int __init cs4270_init(void)
 {
 	return snd_soc_register_dai(&cs4270_dai);
 }
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index f333e88ee25..9a3e67e5319 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -195,7 +195,7 @@ struct snd_soc_codec_device soc_codec_dev_pcm3008 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008);
 
-static int __devinit pcm3008_init(void)
+static int __init pcm3008_init(void)
 {
 	return snd_soc_register_dai(&pcm3008_dai);
 }
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 77fdcb4b9a1..2325aefea41 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -793,7 +793,7 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
 
-static int __devinit ssm2602_modinit(void)
+static int __init ssm2602_modinit(void)
 {
 	return snd_soc_register_dai(&ssm2602_dai);
 }
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index eac449b92bd..39f5b981d25 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -847,7 +847,7 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
 
-static int __devinit tlv320aic23_modinit(void)
+static int __init tlv320aic23_modinit(void)
 {
 	return snd_soc_register_dai(&tlv320aic23_dai);
 }
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index ccd57596186..8da9e5d2e2f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1411,7 +1411,7 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
 
-static int __devinit aic3x_modinit(void)
+static int __init aic3x_modinit(void)
 {
 	return snd_soc_register_dai(&aic3x_dai);
 }
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 8e035b5d733..a2c5064a774 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -651,7 +651,7 @@ struct snd_soc_codec_device soc_codec_dev_uda134x = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
 
-static int __devinit uda134x_init(void)
+static int __init uda134x_init(void)
 {
 	return snd_soc_register_dai(&uda134x_dai);
 }
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 55a99b6a68a..e6bf0844fbf 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -841,7 +841,7 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
 
-static int __devinit uda1380_modinit(void)
+static int __init uda1380_modinit(void)
 {
 	return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
 }
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index a2af04bb4e9..40f8238df71 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -889,7 +889,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
 
-static int __devinit wm8510_modinit(void)
+static int __init wm8510_modinit(void)
 {
 	return snd_soc_register_dai(&wm8510_dai);
 }
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 391ec2978ae..d004e584529 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -1042,7 +1042,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
 
-static int __devinit wm8580_modinit(void)
+static int __init wm8580_modinit(void)
 {
 	return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
 }
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index d905e25b1a9..80b11983e13 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -568,7 +568,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8728 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728);
 
-static int __devinit wm8728_modinit(void)
+static int __init wm8728_modinit(void)
 {
 	return snd_soc_register_dai(&wm8728_dai);
 }
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7b455a60d71..c444b9f2701 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -793,7 +793,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
-static int __devinit wm8731_modinit(void)
+static int __init wm8731_modinit(void)
 {
 	return snd_soc_register_dai(&wm8731_dai);
 }
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 84a6307de90..5997fa68e0d 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -1085,7 +1085,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
 
-static int __devinit wm8750_modinit(void)
+static int __init wm8750_modinit(void)
 {
 	return snd_soc_register_dai(&wm8750_dai);
 }
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 1caca30d081..6c21b50c937 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1874,7 +1874,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
 
-static int __devinit wm8753_modinit(void)
+static int __init wm8753_modinit(void)
 {
 	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
 }
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 34e58af0c65..ebf58fba1be 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1473,7 +1473,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
 
-static int __devinit wm8900_modinit(void)
+static int __init wm8900_modinit(void)
 {
 	return i2c_add_driver(&wm8900_i2c_driver);
 }
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 5d8fe7e1571..0b5bea37e3d 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1809,7 +1809,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903);
 
-static int __devinit wm8903_modinit(void)
+static int __init wm8903_modinit(void)
 {
 	return snd_soc_register_dai(&wm8903_dai);
 }
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 2979fc4f44f..88ead7f8dd9 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -935,7 +935,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = {
 
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
 
-static int __devinit wm8971_modinit(void)
+static int __init wm8971_modinit(void)
 {
 	return snd_soc_register_dai(&wm8971_dai);
 }
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 53e71aafe6c..5b5afc14447 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1643,7 +1643,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
 
-static int __devinit wm8990_modinit(void)
+static int __init wm8990_modinit(void)
 {
 	return snd_soc_register_dai(&wm8990_dai);
 }
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index d89fc2f009a..81ff5c37ab5 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -481,7 +481,7 @@ struct snd_soc_dai davinci_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
-static int __devinit davinci_i2s_init(void)
+static int __init davinci_i2s_init(void)
 {
 	return snd_soc_register_dai(&davinci_i2s_dai);
 }
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index f1b6e02d24e..bc83e1cbd17 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -384,7 +384,7 @@ struct snd_soc_platform davinci_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(davinci_soc_platform);
 
-static int __devinit davinci_soc_platform_init(void)
+static int __init davinci_soc_platform_init(void)
 {
 	return snd_soc_register_platform(&davinci_soc_platform);
 }
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 646c807163a..64993eda567 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -853,7 +853,7 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info)
 }
 EXPORT_SYMBOL_GPL(fsl_dma_configure);
 
-static int __devinit fsl_soc_platform_init(void)
+static int __init fsl_soc_platform_init(void)
 {
 	return snd_soc_register_platform(&fsl_soc_platform);
 }
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 41cab203416..39cc57ce4bf 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -499,7 +499,7 @@ struct snd_soc_dai omap_mcbsp_dai[] = {
 
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
-static int __devinit omap_mcbsp_init(void)
+static int __init omap_mcbsp_init(void)
 {
 	return snd_soc_register_dais(omap_mcbsp_dai,
 				     ARRAY_SIZE(omap_mcbsp_dai));
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 9940de29631..ce580a97cbd 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -354,7 +354,7 @@ struct snd_soc_platform omap_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(omap_soc_platform);
 
-static int __devinit omap_soc_platform_init(void)
+static int __init omap_soc_platform_init(void)
 {
 	return snd_soc_register_platform(&omap_soc_platform);
 }
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 3587f2fae5f..73cb6b4c2f2 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -913,7 +913,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 };
 EXPORT_SYMBOL_GPL(pxa_ssp_dai);
 
-static int __devinit pxa_ssp_init(void)
+static int __init pxa_ssp_init(void)
 {
 	return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
 }
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index f6249d5b492..780db6757ad 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -228,7 +228,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
-static int __devinit pxa_ac97_init(void)
+static int __init pxa_ac97_init(void)
 {
 	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
 }
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 4fa1578f5d4..c670d08e7c9 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -118,7 +118,7 @@ struct snd_soc_platform pxa2xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
 
-static int __devinit pxa2xx_soc_platform_init(void)
+static int __init pxa2xx_soc_platform_init(void)
 {
 	return snd_soc_register_platform(&pxa2xx_soc_platform);
 }
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 2cf05079156..f3fc0aba0aa 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -736,7 +736,7 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
-static int __devinit s3c2412_i2s_init(void)
+static int __init s3c2412_i2s_init(void)
 {
 	return snd_soc_register_dai(&s3c2412_i2s_dai);
 }
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 7360c6d913b..723779a3058 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -396,7 +396,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
-static int __devinit s3c2443_ac97_init(void)
+static int __init s3c2443_ac97_init(void)
 {
 	return snd_soc_register_dai(&s3c2443_ac97_dai);
 }
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 897b1ac92ce..6f4d439b57a 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -482,7 +482,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
-static int __devinit s3c24xx_i2s_init(void)
+static int __init s3c24xx_i2s_init(void)
 {
 	return snd_soc_register_dai(&s3c24xx_i2s_dai);
 }
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index ea5a9caec13..7c64d31d067 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -465,7 +465,7 @@ struct snd_soc_platform s3c24xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
 
-static int __devinit s3c24xx_soc_platform_init(void)
+static int __init s3c24xx_soc_platform_init(void)
 {
 	return snd_soc_register_platform(&s3c24xx_soc_platform);
 }
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 39ffca0933a..0dad3a0bb92 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -348,7 +348,7 @@ struct snd_soc_platform sh7760_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(sh7760_soc_platform);
 
-static int __devinit sh7760_soc_platform_init(void)
+static int __init sh7760_soc_platform_init(void)
 {
 	return snd_soc_register_platform(&sh7760_soc_platform);
 }
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index 9169bad1acf..eab31838bad 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -314,7 +314,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_hac_dai);
 
-static int __devinit sh4_hac_init(void)
+static int __init sh4_hac_init(void)
 {
 	return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
 }
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 9093588d4d0..d1e5390fdde 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -392,7 +392,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_ssi_dai);
 
-static int __devinit sh4_ssi_init(void)
+static int __init sh4_ssi_init(void)
 {
 	return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
 }
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 76a89eb65ba..4d2db7cfaf4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2224,7 +2224,7 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform)
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
 
-static int __devinit snd_soc_init(void)
+static int __init snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
 	debugfs_root = debugfs_create_dir("asoc", NULL);
-- 
cgit v1.2.3-70-g09d2