sound/soc/imx/wm1133-ev1.c - maze/linux - Git at Google

 /*
  *  wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
  *
  *  Copyright (c) 2010 Wolfson Microelectronics plc
  *  Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
  *  Based on an earlier driver for the same hardware by Liam Girdwood.
  *
  *  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/platform_device.h>
 #include <linux/clk.h>
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/jack.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>

 #include <mach/audmux.h>

 #include "imx-ssi.h"
 #include "../codecs/wm8350.h"

 /* There is a silicon mic on the board optionally connected via a solder pad
  * SP1.  Define this to enable it.
  */
 #undef USE_SIMIC

 struct _wm8350_audio {
 	unsigned int channels;
 	snd_pcm_format_t format;
 	unsigned int rate;
 	unsigned int sysclk;
 	unsigned int bclkdiv;
 	unsigned int clkdiv;
 	unsigned int lr_rate;
 };

 /* in order of power consumption per rate (lowest first) */
 static const struct _wm8350_audio wm8350_audio[] = {
 	/* 16bit mono modes */
 	{1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
 	 WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},

 	/* 16 bit stereo modes */
 	{2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
 	 WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
 	 WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
 	 WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
 	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
 	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
 	 WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
 	 WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
 	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
 	{2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
 	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},

 	/* 24bit stereo modes */
 	{2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
 	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
 	{2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
 	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
 	{2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
 	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
 	{2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
 	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
 };

 static int wm1133_ev1_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->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int i, found = 0;
 	snd_pcm_format_t format = params_format(params);
 	unsigned int rate = params_rate(params);
 	unsigned int channels = params_channels(params);
 	u32 dai_format;

 	/* find the correct audio parameters */
 	for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
 		if (rate == wm8350_audio[i].rate &&
 		    format == wm8350_audio[i].format &&
 		    channels == wm8350_audio[i].channels) {
 			found = 1;
 			break;
 		}
 	}
 	if (!found)
 		return -EINVAL;

 	/* codec FLL input is 14.75 MHz from MCLK */
 	snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);

 	dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 		SND_SOC_DAIFMT_CBM_CFM;

 	/* set codec DAI configuration */
 	snd_soc_dai_set_fmt(codec_dai, dai_format);

 	/* set cpu DAI configuration */
 	snd_soc_dai_set_fmt(cpu_dai, dai_format);

 	/* TODO: The SSI driver should figure this out for us */
 	switch (channels) {
 	case 2:
 		snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
 		break;
 	case 1:
 		snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
 		break;
 	default:
 		return -EINVAL;
 	}

 	/* set MCLK as the codec system clock for DAC and ADC */
 	snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
 			       wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);

 	/* set codec BCLK division for sample rate */
 	snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
 			       wm8350_audio[i].bclkdiv);

 	/* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
 	snd_soc_dai_set_clkdiv(codec_dai,
 			       WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
 	snd_soc_dai_set_clkdiv(codec_dai,
 			       WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);

 	/* now configure DAC and ADC clocks */
 	snd_soc_dai_set_clkdiv(codec_dai,
 			       WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);

 	snd_soc_dai_set_clkdiv(codec_dai,
 			       WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);

 	return 0;
 }

 static struct snd_soc_ops wm1133_ev1_ops = {
 	.hw_params = wm1133_ev1_hw_params,
 };

 static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
 #ifdef USE_SIMIC
 	SND_SOC_DAPM_MIC("SiMIC", NULL),
 #endif
 	SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
 	SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
 	SND_SOC_DAPM_LINE("Line In Jack", NULL),
 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 };

 /* imx32ads soc_card audio map */
 static const struct snd_soc_dapm_route wm1133_ev1_map[] = {

 #ifdef USE_SIMIC
 	/* SiMIC --> IN1LN (with automatic bias) via SP1 */
 	{ "IN1LN", NULL, "Mic Bias" },
 	{ "Mic Bias", NULL, "SiMIC" },
 #endif

 	/* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
 	{ "IN1LN", NULL, "Mic Bias" },
 	{ "IN1LP", NULL, "Mic1 Jack" },
 	{ "Mic Bias", NULL, "Mic1 Jack" },

 	/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
 	{ "IN1RN", NULL, "Mic Bias" },
 	{ "IN1RP", NULL, "Mic2 Jack" },
 	{ "Mic Bias", NULL, "Mic2 Jack" },

 	/* Line in Jack --> AUX (L+R) */
 	{ "IN3R", NULL, "Line In Jack" },
 	{ "IN3L", NULL, "Line In Jack" },

 	/* Out1 --> Headphone Jack */
 	{ "Headphone Jack", NULL, "OUT1R" },
 	{ "Headphone Jack", NULL, "OUT1L" },

 	/* Out1 --> Line Out Jack */
 	{ "Line Out Jack", NULL, "OUT2R" },
 	{ "Line Out Jack", NULL, "OUT2L" },
 };

 static struct snd_soc_jack hp_jack;

 static struct snd_soc_jack_pin hp_jack_pins[] = {
 	{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
 };

 static struct snd_soc_jack mic_jack;

 static struct snd_soc_jack_pin mic_jack_pins[] = {
 	{ .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
 	{ .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
 };

 static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;

 	snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
 				  ARRAY_SIZE(wm1133_ev1_widgets));

 	snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
 				ARRAY_SIZE(wm1133_ev1_map));

 	/* Headphone jack detection */
 	snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
 	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
 			      hp_jack_pins);
 	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);

 	/* Microphone jack detection */
 	snd_soc_jack_new(codec, "Microphone",
 			 SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
 	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
 			      mic_jack_pins);
 	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
 			       SND_JACK_BTN_0);

 	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");

 	return 0;
 }


 static struct snd_soc_dai_link wm1133_ev1_dai = {
 	.name = "WM1133-EV1",
 	.stream_name = "Audio",
 	.cpu_dai_name = "imx-ssi.0",
 	.codec_dai_name = "wm8350-hifi",
 	.platform_name = "imx-fiq-pcm-audio.0",
 	.codec_name = "wm8350-codec.0-0x1a",
 	.init = wm1133_ev1_init,
 	.ops = &wm1133_ev1_ops,
 	.symmetric_rates = 1,
 };

 static struct snd_soc_card wm1133_ev1 = {
 	.name = "WM1133-EV1",
 	.dai_link = &wm1133_ev1_dai,
 	.num_links = 1,
 };

 static struct platform_device *wm1133_ev1_snd_device;

 static int __init wm1133_ev1_audio_init(void)
 {
 	int ret;
 	unsigned int ptcr, pdcr;

 	/* SSI0 mastered by port 5 */
 	ptcr = MXC_AUDMUX_V2_PTCR_SYN |
 		MXC_AUDMUX_V2_PTCR_TFSDIR |
 		MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
 		MXC_AUDMUX_V2_PTCR_TCLKDIR |
 		MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
 	pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
 	mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);

 	ptcr = MXC_AUDMUX_V2_PTCR_SYN;
 	pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
 	mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);

 	wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!wm1133_ev1_snd_device)
 		return -ENOMEM;

 	platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1);
 	ret = platform_device_add(wm1133_ev1_snd_device);

 	if (ret)
 		platform_device_put(wm1133_ev1_snd_device);

 	return ret;
 }
 module_init(wm1133_ev1_audio_init);

 static void __exit wm1133_ev1_audio_exit(void)
 {
 	platform_device_unregister(wm1133_ev1_snd_device);
 }
 module_exit(wm1133_ev1_audio_exit);

 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
 MODULE_LICENSE("GPL");
	/*
	* wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
	*
	* Copyright (c) 2010 Wolfson Microelectronics plc
	* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
	*
	* Based on an earlier driver for the same hardware by Liam Girdwood.
	*
	* 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/platform_device.h>
	#include <linux/clk.h>
	#include <linux/module.h>
	#include <sound/core.h>
	#include <sound/jack.h>
	#include <sound/pcm.h>
	#include <sound/pcm_params.h>
	#include <sound/soc.h>

	#include <mach/audmux.h>

	#include "imx-ssi.h"
	#include "../codecs/wm8350.h"

	/* There is a silicon mic on the board optionally connected via a solder pad
	* SP1. Define this to enable it.
	*/
	#undef USE_SIMIC

	struct _wm8350_audio {
	unsigned int channels;
	snd_pcm_format_t format;
	unsigned int rate;
	unsigned int sysclk;
	unsigned int bclkdiv;
	unsigned int clkdiv;
	unsigned int lr_rate;
	};

	/* in order of power consumption per rate (lowest first) */
	static const struct _wm8350_audio wm8350_audio[] = {
	/* 16bit mono modes */
	{1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
	WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},

	/* 16 bit stereo modes */
	{2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
	WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
	WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
	WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
	WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
	WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
	WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
	WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
	WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
	{2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
	WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},

	/* 24bit stereo modes */
	{2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
	WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
	{2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
	WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
	{2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
	WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
	{2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
	WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
	};

	static int wm1133_ev1_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->codec_dai;
	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
	int i, found = 0;
	snd_pcm_format_t format = params_format(params);
	unsigned int rate = params_rate(params);
	unsigned int channels = params_channels(params);
	u32 dai_format;

	/* find the correct audio parameters */
	for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
	if (rate == wm8350_audio[i].rate &&
	format == wm8350_audio[i].format &&
	channels == wm8350_audio[i].channels) {
	found = 1;
	break;
	}
	}
	if (!found)
	return -EINVAL;

	/* codec FLL input is 14.75 MHz from MCLK */
	snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);

	dai_format = SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \|
	SND_SOC_DAIFMT_CBM_CFM;

	/* set codec DAI configuration */
	snd_soc_dai_set_fmt(codec_dai, dai_format);

	/* set cpu DAI configuration */
	snd_soc_dai_set_fmt(cpu_dai, dai_format);

	/* TODO: The SSI driver should figure this out for us */
	switch (channels) {
	case 2:
	snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
	break;
	case 1:
	snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
	break;
	default:
	return -EINVAL;
	}

	/* set MCLK as the codec system clock for DAC and ADC */
	snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
	wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);

	/* set codec BCLK division for sample rate */
	snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
	wm8350_audio[i].bclkdiv);

	/* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
	snd_soc_dai_set_clkdiv(codec_dai,
	WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
	snd_soc_dai_set_clkdiv(codec_dai,
	WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);

	/* now configure DAC and ADC clocks */
	snd_soc_dai_set_clkdiv(codec_dai,
	WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);

	snd_soc_dai_set_clkdiv(codec_dai,
	WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);

	return 0;
	}

	static struct snd_soc_ops wm1133_ev1_ops = {
	.hw_params = wm1133_ev1_hw_params,
	};

	static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
	#ifdef USE_SIMIC
	SND_SOC_DAPM_MIC("SiMIC", NULL),
	#endif
	SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
	SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
	SND_SOC_DAPM_LINE("Line In Jack", NULL),
	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
	SND_SOC_DAPM_HP("Headphone Jack", NULL),
	};

	/* imx32ads soc_card audio map */
	static const struct snd_soc_dapm_route wm1133_ev1_map[] = {

	#ifdef USE_SIMIC
	/* SiMIC --> IN1LN (with automatic bias) via SP1 */
	{ "IN1LN", NULL, "Mic Bias" },
	{ "Mic Bias", NULL, "SiMIC" },
	#endif

	/* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
	{ "IN1LN", NULL, "Mic Bias" },
	{ "IN1LP", NULL, "Mic1 Jack" },
	{ "Mic Bias", NULL, "Mic1 Jack" },

	/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
	{ "IN1RN", NULL, "Mic Bias" },
	{ "IN1RP", NULL, "Mic2 Jack" },
	{ "Mic Bias", NULL, "Mic2 Jack" },

	/* Line in Jack --> AUX (L+R) */
	{ "IN3R", NULL, "Line In Jack" },
	{ "IN3L", NULL, "Line In Jack" },

	/* Out1 --> Headphone Jack */
	{ "Headphone Jack", NULL, "OUT1R" },
	{ "Headphone Jack", NULL, "OUT1L" },

	/* Out1 --> Line Out Jack */
	{ "Line Out Jack", NULL, "OUT2R" },
	{ "Line Out Jack", NULL, "OUT2L" },
	};

	static struct snd_soc_jack hp_jack;

	static struct snd_soc_jack_pin hp_jack_pins[] = {
	{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
	};

	static struct snd_soc_jack mic_jack;

	static struct snd_soc_jack_pin mic_jack_pins[] = {
	{ .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
	{ .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
	};

	static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
	{
	struct snd_soc_codec *codec = rtd->codec;
	struct snd_soc_dapm_context *dapm = &codec->dapm;

	snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
	ARRAY_SIZE(wm1133_ev1_widgets));

	snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
	ARRAY_SIZE(wm1133_ev1_map));

	/* Headphone jack detection */
	snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
	hp_jack_pins);
	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);

	/* Microphone jack detection */
	snd_soc_jack_new(codec, "Microphone",
	SND_JACK_MICROPHONE \| SND_JACK_BTN_0, &mic_jack);
	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
	mic_jack_pins);
	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
	SND_JACK_BTN_0);

	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");

	return 0;
	}


	static struct snd_soc_dai_link wm1133_ev1_dai = {
	.name = "WM1133-EV1",
	.stream_name = "Audio",
	.cpu_dai_name = "imx-ssi.0",
	.codec_dai_name = "wm8350-hifi",
	.platform_name = "imx-fiq-pcm-audio.0",
	.codec_name = "wm8350-codec.0-0x1a",
	.init = wm1133_ev1_init,
	.ops = &wm1133_ev1_ops,
	.symmetric_rates = 1,
	};

	static struct snd_soc_card wm1133_ev1 = {
	.name = "WM1133-EV1",
	.dai_link = &wm1133_ev1_dai,
	.num_links = 1,
	};

	static struct platform_device *wm1133_ev1_snd_device;

	static int __init wm1133_ev1_audio_init(void)
	{
	int ret;
	unsigned int ptcr, pdcr;

	/* SSI0 mastered by port 5 */
	ptcr = MXC_AUDMUX_V2_PTCR_SYN \|
	MXC_AUDMUX_V2_PTCR_TFSDIR \|
	MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) \|
	MXC_AUDMUX_V2_PTCR_TCLKDIR \|
	MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
	pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
	mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);

	ptcr = MXC_AUDMUX_V2_PTCR_SYN;
	pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
	mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);

	wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
	if (!wm1133_ev1_snd_device)
	return -ENOMEM;

	platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1);
	ret = platform_device_add(wm1133_ev1_snd_device);

	if (ret)
	platform_device_put(wm1133_ev1_snd_device);

	return ret;
	}
	module_init(wm1133_ev1_audio_init);

	static void __exit wm1133_ev1_audio_exit(void)
	{
	platform_device_unregister(wm1133_ev1_snd_device);
	}
	module_exit(wm1133_ev1_audio_exit);

	MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
	MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
	MODULE_LICENSE("GPL");