sound/soc/ambarella/i1evk.c - nest-cam/v179/linux - Git at Google

 /*
  * sound/soc/i1sevk.c
  *
  * Author: Cao Rongrong <rrcao@ambarella.com>
  *
  * History:
  *	2011/03/28 - [Cao Rongrong] Created file
  *
  * Copyright (C) 2004-2009, Ambarella, Inc.
  *
  * 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.
  *
  * This program is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  * GNU General Public License for more details.
  *
  * You should have received a copy of the GNU General Public License
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
  *
  */

 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/io.h>
 #include <linux/platform_device.h>
 #include <sound/soc.h>
 #include <sound/jack.h>
 #include <asm/mach-types.h>
 #include <mach/gpio.h>
 #include <plat/audio.h>

 #include "amdroid_jack.h"
 #include "ambarella_i2s.h"
 #include <linux/mfd/wm8994/core.h>
 #include <linux/mfd/wm8994/registers.h>
 #include "../codecs/wm8994.h"

 /* If don't use alsa standard Jack detction mechanism,
  * switch class will take over this job. */
 static unsigned int gpio_jack = 1;
 module_param(gpio_jack, uint, S_IRUGO);
 MODULE_PARM_DESC(gpio_jack, "Whether or not use alsa standard Jack detection mechanism.");

 /* Headset jack */
 static struct snd_soc_jack hs_jack;

 /* Headset jack detection DAPM pins */
 static struct snd_soc_jack_pin hs_jack_pins[] = {
 	{
 		.pin = "Headset Mic",
 		.mask = SND_JACK_MICROPHONE,
 	},
 	{
 		.pin = "Headset Stereophone",
 		.mask = SND_JACK_HEADPHONE,
 	},
 };

 /* Headset jack detection gpios */
 static struct snd_soc_jack_gpio hs_jack_gpios[] = {
 	{
 		.gpio = GPIO(12),
 		.name = "hsdet-gpio",
 		.report = SND_JACK_HEADSET,
 		.debounce_time = 200,
 	},
 };

 /* AV jack */
 static struct snd_soc_jack av_jack;

 /* AV jack detection DAPM pins */
 static struct snd_soc_jack_pin av_jack_pins[] = {
 	{
 		.pin = "Line Out 2",
 		.mask = SND_JACK_LINEOUT,
 	},
 };

 /* AV jack detection gpios */
 static struct snd_soc_jack_gpio av_jack_gpios[] = {
 	{
 		.gpio = GPIO(13),
 		.name = "avdet-gpio",
 		.report = SND_JACK_LINEOUT,
 		.debounce_time = 200,
 	},
 };

 static struct amdroid_jack_zone amdroid_jack_zones[] = {
 	{
 		/* 0 <= adc <= 0x200, unstable zone, default to 3pole if it stays
 		 * in this range for a half second (20ms delays, 25 samples) */
 		.adc_high = 0x200,
 		.delay_ms = 20,
 		.check_count = 25,
 		.jack_type = AMDROID_HEADSET_3POLE,
 	},
 	{
 		/* 200 < adc, 4 pole zone, default to 4pole if it stays
 		 * in this range for half second (20ms delays, 10 samples) */
 		.adc_high = 0x7fffffff,
 		.delay_ms = 20,
 		.check_count = 25,
 		.jack_type = AMDROID_HEADSET_4POLE,
 	},
 };

 static void platform_amdroid_jack_release(struct device * dev)
 {
 	return ;
 }

 static void jack_set_micbias_state(void *private_data, bool on)
 {
 	struct snd_soc_dapm_context *dapm = private_data;

 	if (on)
 		snd_soc_dapm_force_enable_pin(dapm, "MICBIAS2");
 	else
 		snd_soc_dapm_disable_pin(dapm, "MICBIAS2");

 	snd_soc_dapm_sync(dapm);
 }

 struct amdroid_jack_platform_data amdroid_jack_data = {
 	.set_micbias_state = jack_set_micbias_state,
 	.adc_channel = 0,
 	.zones = amdroid_jack_zones,
 	.num_zones = ARRAY_SIZE(amdroid_jack_zones),
 	.detect_gpio = GPIO(12),
 	.active_high = 1,
 };

 static struct platform_device amdroid_jack_device = {
        .name             = "amdroid_jack",
        .dev = {
                .platform_data = &amdroid_jack_data,
 	       .release	= platform_amdroid_jack_release,
        }
 };

 static const struct snd_soc_dapm_widget i1evk_dapm_widgets[] = {
 	/* Output */
 	SND_SOC_DAPM_SPK("Ext Left Spk", NULL),
 	SND_SOC_DAPM_SPK("Ext Right Spk", NULL),
 	SND_SOC_DAPM_SPK("Earphone", NULL),
 	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
 	SND_SOC_DAPM_LINE("Line Out 1", NULL),
 	SND_SOC_DAPM_LINE("Line Out 2", NULL),
 	/* Input */
 	SND_SOC_DAPM_MIC("Main Mic", NULL),
 	SND_SOC_DAPM_MIC("2nd Mic", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_LINE("FM Left In", NULL),
 	SND_SOC_DAPM_LINE("FM Right In", NULL),
 	SND_SOC_DAPM_LINE("3G In", NULL),
 };

 static const struct snd_soc_dapm_route i1evk_dapm_routes[] = {
 	{"Ext Left Spk", NULL, "SPKOUTLP"},
 	{"Ext Left Spk", NULL, "SPKOUTLN"},

 	{"Ext Right Spk", NULL, "SPKOUTRP"},
 	{"Ext Right Spk", NULL, "SPKOUTRN"},

 	{"Earphone", NULL, "HPOUT2N"},
 	{"Earphone", NULL, "HPOUT2P"},

 	{"Headset Stereophone", NULL, "HPOUT1L"},
 	{"Headset Stereophone", NULL, "HPOUT1R"},

 	{"Line Out 1", NULL, "LINEOUT1N"},
 	{"Line Out 1", NULL, "LINEOUT1P"},

 	{"Line Out 2", NULL, "LINEOUT2N"},
 	{"Line Out 2", NULL, "LINEOUT2P"},

 	{"IN1LN", NULL, "MICBIAS1"},
 	{"MICBIAS1", NULL, "Main Mic"},

 	{"IN1LP", NULL, "FM Left In"},

 	{"IN2LN", NULL, "MICBIAS2"},
 	{"MICBIAS2", NULL, "Headset Mic"},

 	{"IN1RN", NULL, "MICBIAS1"},
 	{"MICBIAS1", NULL, "2nd Mic"},

 	{"IN1RP", NULL, "FM Right In"},

 	{"IN2LP:VXRN", NULL, "3G In"},
 	{"IN2RP:VXRP", NULL, "3G In"},
 };

 static int i1evk_wm8994_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int errorCode = 0;

 	/* add i1evk specific widgets */
 	snd_soc_dapm_new_controls(dapm, i1evk_dapm_widgets,
 			ARRAY_SIZE(i1evk_dapm_widgets));

 	/* set up i1evk specific audio routes */
 	snd_soc_dapm_add_routes(dapm, i1evk_dapm_routes,
 			ARRAY_SIZE(i1evk_dapm_routes));

 	snd_soc_dapm_sync(dapm);

 	ambarella_i2s_add_controls(codec);

 	if (gpio_jack) {
 		/* Headset jack detection */
 		errorCode = snd_soc_jack_new(codec,
 				"Headset Jack", SND_JACK_HEADSET, &hs_jack);
 		if (errorCode)
 			return errorCode;

 		errorCode = snd_soc_jack_add_pins(&hs_jack,
 				ARRAY_SIZE(hs_jack_pins), hs_jack_pins);
 		if (errorCode)
 			return errorCode;

 		errorCode = snd_soc_jack_add_gpios(&hs_jack,
 				ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios);
 		if (errorCode)
 			return errorCode;

 		/* Headset jack detection */
 		errorCode = snd_soc_jack_new(codec,
 				"AV Jack", SND_JACK_LINEOUT, &av_jack);
 		if (errorCode)
 			return errorCode;

 		errorCode = snd_soc_jack_add_pins(&av_jack,
 				ARRAY_SIZE(av_jack_pins), av_jack_pins);
 		if (errorCode)
 			return errorCode;

 		errorCode = snd_soc_jack_add_gpios(&av_jack,
 				ARRAY_SIZE(av_jack_gpios), av_jack_gpios);
 	} else {
 		amdroid_jack_data.private_data = dapm;
 		platform_device_register(&amdroid_jack_device);
 	}

 	return errorCode;
 }

 static int i1evk_hifi_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 errorCode = 0, amb_mclk, mclk, oversample;

 	switch (params_rate(params)) {
 	case 8000:
 		amb_mclk = AudioCodec_4_096M;
 		mclk = 4096000;
 		oversample = AudioCodec_512xfs;
 		break;
 	case 11025:
 		amb_mclk = AudioCodec_5_6448M;
 		mclk = 5644800;
 		oversample = AudioCodec_512xfs;
 		break;
 	case 16000:
 		amb_mclk = AudioCodec_4_096M;
 		mclk = 4096000;
 		oversample = AudioCodec_256xfs;
 		break;
 	case 22050:
 		amb_mclk = AudioCodec_5_6448M;
 		mclk = 5644800;
 		oversample = AudioCodec_256xfs;
 		break;
 	case 24000:
 		amb_mclk = AudioCodec_6_144;
 		mclk = 6144000;
 		oversample = AudioCodec_256xfs;
 		break;
 	case 32000:
 		amb_mclk = AudioCodec_8_192M;
 		mclk = 8192000;
 		oversample = AudioCodec_256xfs;
 		break;
 	case 44100:
 		amb_mclk = AudioCodec_11_2896M;
 		mclk = 11289600;
 		oversample = AudioCodec_256xfs;
 		break;
 	case 48000:
 		amb_mclk = AudioCodec_12_288M;
 		mclk = 12288000;
 		oversample = AudioCodec_256xfs;
 		break;
 	default:
 		errorCode = -EINVAL;
 		goto hifi_hw_params_exit;
 	}

 	/* set the I2S system data format*/
 	errorCode = snd_soc_dai_set_fmt(codec_dai,
 		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (errorCode < 0) {
 		pr_err("can't set codec DAI configuration\n");
 		goto hifi_hw_params_exit;
 	}

 	errorCode = snd_soc_dai_set_fmt(cpu_dai,
 		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (errorCode < 0) {
 		pr_err("can't set cpu DAI configuration\n");
 		goto hifi_hw_params_exit;
 	}

 	/* set the I2S system clock*/
 	errorCode = snd_soc_dai_set_sysclk(cpu_dai, AMBARELLA_CLKSRC_ONCHIP, amb_mclk, 0);
 	if (errorCode < 0) {
 		pr_err("can't set cpu MCLK configuration\n");
 		goto hifi_hw_params_exit;
 	}

 	errorCode = snd_soc_dai_set_clkdiv(cpu_dai, AMBARELLA_CLKDIV_LRCLK, oversample);
 	if (errorCode < 0) {
 		pr_err("can't set cpu MCLK/SF ratio\n");
 		goto hifi_hw_params_exit;
 	}
 #if 0
 	errorCode = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
 					mclk, params_rate(params) * 256);
 	if (errorCode < 0) {
 		pr_err("can't set codec pll configuration\n");
 		goto hifi_hw_params_exit;
 	}

 	errorCode = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
 					params_rate(params) * 256, SND_SOC_CLOCK_IN);
 	if (errorCode < 0) {
 		pr_err("can't set codec MCLK configuration\n");
 		goto hifi_hw_params_exit;
 	}
 #else
 	errorCode = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, mclk, 0);
 	if (errorCode < 0) {
 		pr_err("can't set cpu MCLK configuration\n");
 		goto hifi_hw_params_exit;
 	}

 #endif

 hifi_hw_params_exit:
 	return errorCode;
 }

 static struct snd_soc_ops i1evk_hifi_ops = {
 	.hw_params = i1evk_hifi_hw_params,
 };

 static int i1evk_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->codec_dai;
 	int errorCode = 0;

 	if (params_rate(params) != 8000) {
 		pr_err("Voice dai only support 8000Hz sample rate!\n");
 		return -EINVAL;
 	}

 	errorCode = snd_soc_dai_set_fmt(codec_dai,
 		SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (errorCode < 0) {
 		pr_err("can't set codec pll configuration\n");
 		goto voice_hw_params_exit;
 	}

 #if 0	/* When enable these codes, we need to specify BCLK2_FREQ. */
 	/* Set FLL2, use BCLK2 as its source */
 	errorCode = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, WM8994_FLL_SRC_BCLK,
 					BCLK2_FREQ, params_rate(params) * 512);
 	if (errorCode < 0) {
 		pr_err("can't set codec FLL2 configuration\n");
 		goto voice_hw_params_exit;
 	}

 	errorCode = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2,
 					params_rate(params) * 512, SND_SOC_CLOCK_IN);
 	if (errorCode < 0) {
 		pr_err("can't set codec SYSCLK configuration\n");
 		goto voice_hw_params_exit;
 	}
 #endif
 	return 0;

 voice_hw_params_exit:
 	return errorCode;
 }

 static struct snd_soc_ops i1evk_voice_ops = {
 	.hw_params = i1evk_voice_hw_params,
 };

 static struct snd_soc_dai_link i1evk_dai_link[] = {
 	{
 		.name = "WM8994",
 		.stream_name = "WM8994 HiFi",
 		.cpu_dai_name = "ambarella-i2s.0",
 		.codec_dai_name = "wm8994-aif1",
 		.platform_name = "ambarella-pcm-audio",
 		.codec_name = "wm8994-codec",
 		.init = i1evk_wm8994_init,
 		.ops = &i1evk_hifi_ops,
 	},
 	{	/* Bluetooth/3G Voice */
 		.name = "WM8994 Voice",
 		.stream_name = "Voice",
 		.cpu_dai_name = "AMBARELLA_DUMMY_CODEC",
 		.codec_dai_name = "wm8994-aif2",
 		.codec_name = "wm8994-codec",
 		.ops = &i1evk_voice_ops,
 	},
 };

 static struct snd_soc_card snd_soc_card_i1evk = {
 	.name = "I1EVK",
 	.dai_link = i1evk_dai_link,
 	.num_links = ARRAY_SIZE(i1evk_dai_link),
 };


 static struct platform_device *i1evk_snd_device;

 static int __init i1evk_board_init(void)
 {
 	int errorCode = 0;

 	i1evk_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!i1evk_snd_device)
 		return -ENOMEM;

 	platform_set_drvdata(i1evk_snd_device, &snd_soc_card_i1evk);

 	errorCode = platform_device_add(i1evk_snd_device);
 	if (errorCode)
 		goto i1sevk_board_init_exit;

 	return 0;

 i1sevk_board_init_exit:
 	platform_device_put(i1evk_snd_device);
 	return errorCode;
 }

 static void __exit i1evk_board_exit(void)
 {
 	if (gpio_jack) {
 		snd_soc_jack_free_gpios(&hs_jack,
 			ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios);
 		snd_soc_jack_free_gpios(&av_jack,
 			ARRAY_SIZE(av_jack_gpios), av_jack_gpios);
 	} else {
 		platform_device_unregister(&amdroid_jack_device);
 	}

 	platform_device_unregister(i1evk_snd_device);
 }

 module_init(i1evk_board_init);
 module_exit(i1evk_board_exit);

 MODULE_AUTHOR("Cao Rongrong <rrcao@ambarella.com>");
 MODULE_DESCRIPTION("Amabrella i1sevk Board with WM8994 Codec for ALSA");
 MODULE_LICENSE("GPL");
	/*
	* sound/soc/i1sevk.c
	*
	* Author: Cao Rongrong <rrcao@ambarella.com>
	*
	* History:
	* 2011/03/28 - [Cao Rongrong] Created file
	*
	* Copyright (C) 2004-2009, Ambarella, Inc.
	*
	* 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.
	*
	* This program is distributed in the hope that it will be useful,
	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	* GNU General Public License for more details.
	*
	* You should have received a copy of the GNU General Public License
	* along with this program; if not, write to the Free Software
	* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	*
	*/

	#include <linux/module.h>
	#include <linux/moduleparam.h>
	#include <linux/io.h>
	#include <linux/platform_device.h>
	#include <sound/soc.h>
	#include <sound/jack.h>
	#include <asm/mach-types.h>
	#include <mach/gpio.h>
	#include <plat/audio.h>

	#include "amdroid_jack.h"
	#include "ambarella_i2s.h"
	#include <linux/mfd/wm8994/core.h>
	#include <linux/mfd/wm8994/registers.h>
	#include "../codecs/wm8994.h"

	/* If don't use alsa standard Jack detction mechanism,
	* switch class will take over this job. */
	static unsigned int gpio_jack = 1;
	module_param(gpio_jack, uint, S_IRUGO);
	MODULE_PARM_DESC(gpio_jack, "Whether or not use alsa standard Jack detection mechanism.");

	/* Headset jack */
	static struct snd_soc_jack hs_jack;

	/* Headset jack detection DAPM pins */
	static struct snd_soc_jack_pin hs_jack_pins[] = {
	{
	.pin = "Headset Mic",
	.mask = SND_JACK_MICROPHONE,
	},
	{
	.pin = "Headset Stereophone",
	.mask = SND_JACK_HEADPHONE,
	},
	};

	/* Headset jack detection gpios */
	static struct snd_soc_jack_gpio hs_jack_gpios[] = {
	{
	.gpio = GPIO(12),
	.name = "hsdet-gpio",
	.report = SND_JACK_HEADSET,
	.debounce_time = 200,
	},
	};

	/* AV jack */
	static struct snd_soc_jack av_jack;

	/* AV jack detection DAPM pins */
	static struct snd_soc_jack_pin av_jack_pins[] = {
	{
	.pin = "Line Out 2",
	.mask = SND_JACK_LINEOUT,
	},
	};

	/* AV jack detection gpios */
	static struct snd_soc_jack_gpio av_jack_gpios[] = {
	{
	.gpio = GPIO(13),
	.name = "avdet-gpio",
	.report = SND_JACK_LINEOUT,
	.debounce_time = 200,
	},
	};

	static struct amdroid_jack_zone amdroid_jack_zones[] = {
	{
	/* 0 <= adc <= 0x200, unstable zone, default to 3pole if it stays
	* in this range for a half second (20ms delays, 25 samples) */
	.adc_high = 0x200,
	.delay_ms = 20,
	.check_count = 25,
	.jack_type = AMDROID_HEADSET_3POLE,
	},
	{
	/* 200 < adc, 4 pole zone, default to 4pole if it stays
	* in this range for half second (20ms delays, 10 samples) */
	.adc_high = 0x7fffffff,
	.delay_ms = 20,
	.check_count = 25,
	.jack_type = AMDROID_HEADSET_4POLE,
	},
	};

	static void platform_amdroid_jack_release(struct device * dev)
	{
	return ;
	}

	static void jack_set_micbias_state(void *private_data, bool on)
	{
	struct snd_soc_dapm_context *dapm = private_data;

	if (on)
	snd_soc_dapm_force_enable_pin(dapm, "MICBIAS2");
	else
	snd_soc_dapm_disable_pin(dapm, "MICBIAS2");

	snd_soc_dapm_sync(dapm);
	}

	struct amdroid_jack_platform_data amdroid_jack_data = {
	.set_micbias_state = jack_set_micbias_state,
	.adc_channel = 0,
	.zones = amdroid_jack_zones,
	.num_zones = ARRAY_SIZE(amdroid_jack_zones),
	.detect_gpio = GPIO(12),
	.active_high = 1,
	};

	static struct platform_device amdroid_jack_device = {
	.name = "amdroid_jack",
	.dev = {
	.platform_data = &amdroid_jack_data,
	.release = platform_amdroid_jack_release,
	}
	};

	static const struct snd_soc_dapm_widget i1evk_dapm_widgets[] = {
	/* Output */
	SND_SOC_DAPM_SPK("Ext Left Spk", NULL),
	SND_SOC_DAPM_SPK("Ext Right Spk", NULL),
	SND_SOC_DAPM_SPK("Earphone", NULL),
	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
	SND_SOC_DAPM_LINE("Line Out 1", NULL),
	SND_SOC_DAPM_LINE("Line Out 2", NULL),
	/* Input */
	SND_SOC_DAPM_MIC("Main Mic", NULL),
	SND_SOC_DAPM_MIC("2nd Mic", NULL),
	SND_SOC_DAPM_MIC("Headset Mic", NULL),
	SND_SOC_DAPM_LINE("FM Left In", NULL),
	SND_SOC_DAPM_LINE("FM Right In", NULL),
	SND_SOC_DAPM_LINE("3G In", NULL),
	};

	static const struct snd_soc_dapm_route i1evk_dapm_routes[] = {
	{"Ext Left Spk", NULL, "SPKOUTLP"},
	{"Ext Left Spk", NULL, "SPKOUTLN"},

	{"Ext Right Spk", NULL, "SPKOUTRP"},
	{"Ext Right Spk", NULL, "SPKOUTRN"},

	{"Earphone", NULL, "HPOUT2N"},
	{"Earphone", NULL, "HPOUT2P"},

	{"Headset Stereophone", NULL, "HPOUT1L"},
	{"Headset Stereophone", NULL, "HPOUT1R"},

	{"Line Out 1", NULL, "LINEOUT1N"},
	{"Line Out 1", NULL, "LINEOUT1P"},

	{"Line Out 2", NULL, "LINEOUT2N"},
	{"Line Out 2", NULL, "LINEOUT2P"},

	{"IN1LN", NULL, "MICBIAS1"},
	{"MICBIAS1", NULL, "Main Mic"},

	{"IN1LP", NULL, "FM Left In"},

	{"IN2LN", NULL, "MICBIAS2"},
	{"MICBIAS2", NULL, "Headset Mic"},

	{"IN1RN", NULL, "MICBIAS1"},
	{"MICBIAS1", NULL, "2nd Mic"},

	{"IN1RP", NULL, "FM Right In"},

	{"IN2LP:VXRN", NULL, "3G In"},
	{"IN2RP:VXRP", NULL, "3G In"},
	};

	static int i1evk_wm8994_init(struct snd_soc_pcm_runtime *rtd)
	{
	struct snd_soc_codec *codec = rtd->codec;
	struct snd_soc_dapm_context *dapm = &codec->dapm;
	int errorCode = 0;

	/* add i1evk specific widgets */
	snd_soc_dapm_new_controls(dapm, i1evk_dapm_widgets,
	ARRAY_SIZE(i1evk_dapm_widgets));

	/* set up i1evk specific audio routes */
	snd_soc_dapm_add_routes(dapm, i1evk_dapm_routes,
	ARRAY_SIZE(i1evk_dapm_routes));

	snd_soc_dapm_sync(dapm);

	ambarella_i2s_add_controls(codec);

	if (gpio_jack) {
	/* Headset jack detection */
	errorCode = snd_soc_jack_new(codec,
	"Headset Jack", SND_JACK_HEADSET, &hs_jack);
	if (errorCode)
	return errorCode;

	errorCode = snd_soc_jack_add_pins(&hs_jack,
	ARRAY_SIZE(hs_jack_pins), hs_jack_pins);
	if (errorCode)
	return errorCode;

	errorCode = snd_soc_jack_add_gpios(&hs_jack,
	ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios);
	if (errorCode)
	return errorCode;

	/* Headset jack detection */
	errorCode = snd_soc_jack_new(codec,
	"AV Jack", SND_JACK_LINEOUT, &av_jack);
	if (errorCode)
	return errorCode;

	errorCode = snd_soc_jack_add_pins(&av_jack,
	ARRAY_SIZE(av_jack_pins), av_jack_pins);
	if (errorCode)
	return errorCode;

	errorCode = snd_soc_jack_add_gpios(&av_jack,
	ARRAY_SIZE(av_jack_gpios), av_jack_gpios);
	} else {
	amdroid_jack_data.private_data = dapm;
	platform_device_register(&amdroid_jack_device);
	}

	return errorCode;
	}

	static int i1evk_hifi_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 errorCode = 0, amb_mclk, mclk, oversample;

	switch (params_rate(params)) {
	case 8000:
	amb_mclk = AudioCodec_4_096M;
	mclk = 4096000;
	oversample = AudioCodec_512xfs;
	break;
	case 11025:
	amb_mclk = AudioCodec_5_6448M;
	mclk = 5644800;
	oversample = AudioCodec_512xfs;
	break;
	case 16000:
	amb_mclk = AudioCodec_4_096M;
	mclk = 4096000;
	oversample = AudioCodec_256xfs;
	break;
	case 22050:
	amb_mclk = AudioCodec_5_6448M;
	mclk = 5644800;
	oversample = AudioCodec_256xfs;
	break;
	case 24000:
	amb_mclk = AudioCodec_6_144;
	mclk = 6144000;
	oversample = AudioCodec_256xfs;
	break;
	case 32000:
	amb_mclk = AudioCodec_8_192M;
	mclk = 8192000;
	oversample = AudioCodec_256xfs;
	break;
	case 44100:
	amb_mclk = AudioCodec_11_2896M;
	mclk = 11289600;
	oversample = AudioCodec_256xfs;
	break;
	case 48000:
	amb_mclk = AudioCodec_12_288M;
	mclk = 12288000;
	oversample = AudioCodec_256xfs;
	break;
	default:
	errorCode = -EINVAL;
	goto hifi_hw_params_exit;
	}

	/* set the I2S system data format*/
	errorCode = snd_soc_dai_set_fmt(codec_dai,
	SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \| SND_SOC_DAIFMT_CBS_CFS);
	if (errorCode < 0) {
	pr_err("can't set codec DAI configuration\n");
	goto hifi_hw_params_exit;
	}

	errorCode = snd_soc_dai_set_fmt(cpu_dai,
	SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \| SND_SOC_DAIFMT_CBS_CFS);
	if (errorCode < 0) {
	pr_err("can't set cpu DAI configuration\n");
	goto hifi_hw_params_exit;
	}

	/* set the I2S system clock*/
	errorCode = snd_soc_dai_set_sysclk(cpu_dai, AMBARELLA_CLKSRC_ONCHIP, amb_mclk, 0);
	if (errorCode < 0) {
	pr_err("can't set cpu MCLK configuration\n");
	goto hifi_hw_params_exit;
	}

	errorCode = snd_soc_dai_set_clkdiv(cpu_dai, AMBARELLA_CLKDIV_LRCLK, oversample);
	if (errorCode < 0) {
	pr_err("can't set cpu MCLK/SF ratio\n");
	goto hifi_hw_params_exit;
	}
	#if 0
	errorCode = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
	mclk, params_rate(params) * 256);
	if (errorCode < 0) {
	pr_err("can't set codec pll configuration\n");
	goto hifi_hw_params_exit;
	}

	errorCode = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
	params_rate(params) * 256, SND_SOC_CLOCK_IN);
	if (errorCode < 0) {
	pr_err("can't set codec MCLK configuration\n");
	goto hifi_hw_params_exit;
	}
	#else
	errorCode = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, mclk, 0);
	if (errorCode < 0) {
	pr_err("can't set cpu MCLK configuration\n");
	goto hifi_hw_params_exit;
	}

	#endif

	hifi_hw_params_exit:
	return errorCode;
	}

	static struct snd_soc_ops i1evk_hifi_ops = {
	.hw_params = i1evk_hifi_hw_params,
	};

	static int i1evk_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->codec_dai;
	int errorCode = 0;

	if (params_rate(params) != 8000) {
	pr_err("Voice dai only support 8000Hz sample rate!\n");
	return -EINVAL;
	}

	errorCode = snd_soc_dai_set_fmt(codec_dai,
	SND_SOC_DAIFMT_DSP_B \| SND_SOC_DAIFMT_NB_NF \| SND_SOC_DAIFMT_CBS_CFS);
	if (errorCode < 0) {
	pr_err("can't set codec pll configuration\n");
	goto voice_hw_params_exit;
	}

	#if 0 /* When enable these codes, we need to specify BCLK2_FREQ. */
	/* Set FLL2, use BCLK2 as its source */
	errorCode = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, WM8994_FLL_SRC_BCLK,
	BCLK2_FREQ, params_rate(params) * 512);
	if (errorCode < 0) {
	pr_err("can't set codec FLL2 configuration\n");
	goto voice_hw_params_exit;
	}

	errorCode = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2,
	params_rate(params) * 512, SND_SOC_CLOCK_IN);
	if (errorCode < 0) {
	pr_err("can't set codec SYSCLK configuration\n");
	goto voice_hw_params_exit;
	}
	#endif
	return 0;

	voice_hw_params_exit:
	return errorCode;
	}

	static struct snd_soc_ops i1evk_voice_ops = {
	.hw_params = i1evk_voice_hw_params,
	};

	static struct snd_soc_dai_link i1evk_dai_link[] = {
	{
	.name = "WM8994",
	.stream_name = "WM8994 HiFi",
	.cpu_dai_name = "ambarella-i2s.0",
	.codec_dai_name = "wm8994-aif1",
	.platform_name = "ambarella-pcm-audio",
	.codec_name = "wm8994-codec",
	.init = i1evk_wm8994_init,
	.ops = &i1evk_hifi_ops,
	},
	{ /* Bluetooth/3G Voice */
	.name = "WM8994 Voice",
	.stream_name = "Voice",
	.cpu_dai_name = "AMBARELLA_DUMMY_CODEC",
	.codec_dai_name = "wm8994-aif2",
	.codec_name = "wm8994-codec",
	.ops = &i1evk_voice_ops,
	},
	};

	static struct snd_soc_card snd_soc_card_i1evk = {
	.name = "I1EVK",
	.dai_link = i1evk_dai_link,
	.num_links = ARRAY_SIZE(i1evk_dai_link),
	};


	static struct platform_device *i1evk_snd_device;

	static int __init i1evk_board_init(void)
	{
	int errorCode = 0;

	i1evk_snd_device = platform_device_alloc("soc-audio", -1);
	if (!i1evk_snd_device)
	return -ENOMEM;

	platform_set_drvdata(i1evk_snd_device, &snd_soc_card_i1evk);

	errorCode = platform_device_add(i1evk_snd_device);
	if (errorCode)
	goto i1sevk_board_init_exit;

	return 0;

	i1sevk_board_init_exit:
	platform_device_put(i1evk_snd_device);
	return errorCode;
	}

	static void __exit i1evk_board_exit(void)
	{
	if (gpio_jack) {
	snd_soc_jack_free_gpios(&hs_jack,
	ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios);
	snd_soc_jack_free_gpios(&av_jack,
	ARRAY_SIZE(av_jack_gpios), av_jack_gpios);
	} else {
	platform_device_unregister(&amdroid_jack_device);
	}

	platform_device_unregister(i1evk_snd_device);
	}

	module_init(i1evk_board_init);
	module_exit(i1evk_board_exit);

	MODULE_AUTHOR("Cao Rongrong <rrcao@ambarella.com>");
	MODULE_DESCRIPTION("Amabrella i1sevk Board with WM8994 Codec for ALSA");
	MODULE_LICENSE("GPL");