sound/soc/omap/tonka.c - kernel/panda - Git at Google

 /*
  * tonka.c  --  SoC audio for Tonka
  *
  * Author: Misael Lopez Cruz <misael.lopez@ti.com>
  *         Liam Girdwood <lrg@ti.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.
  *
  * 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., 51 Franklin St, Fifth Floor, Boston, MA
  * 02110-1301 USA
  *
  */

 #include <linux/clk.h>
 #include <linux/platform_device.h>
 #include <linux/module.h>

 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/jack.h>
 #include <sound/pcm_params.h>
 #include <sound/soc-dapm.h>
 #include <sound/soc-dsp.h>

 #include <asm/mach-types.h>
 #include <plat/hardware.h>
 #include <plat/mux.h>
 #include <plat/mcbsp.h>

 #include "omap-pcm.h"
 #include "omap-abe.h"
 #include "omap-abe-dsp.h"
 #include "omap-mcbsp.h"

 static int tonka_mcbsp_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 ret = 0;
 	unsigned int channels, div;

 	/* Codec reference clock. */
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN);
 	if (unlikely(ret < 0)) {
 		dev_err(codec_dai->dev, "can't set reference clock\n");
 		return ret;
 	}

 	if (params != NULL) {
 		/* Configure McBSP internal buffer usage */
 		/* this need to be done for playback and/or record */
 		channels = params_channels(params);
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			omap_mcbsp_set_tx_threshold(cpu_dai->id, channels);
 		else
 			omap_mcbsp_set_rx_threshold(cpu_dai->id, channels);
 	}

 	/* Clock master mode; derive bit clock from FCLK. */
 	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK,
 				     24576000, SND_SOC_CLOCK_IN);
 	if (unlikely(ret < 0)) {
 		dev_err(cpu_dai->dev, "can't set reference clock\n");
 		return ret;
 	}

 	switch (params_rate(params)) {
 	case 44100:	/* XXX 45176 */
 		div = 17;
 		break;
 	case 48000:
 		div = 16;
 		break;
 	default:
 		dev_err(cpu_dai->dev, "unknown rate %d\n", params_rate(params));
 		return -EINVAL;
 	}
 	ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, div);
 	if (unlikely(ret < 0)) {
 		dev_err(cpu_dai->dev, "can't set clock divider\n");
 		return ret;
 	}

 	return 0;
 }

 static struct snd_soc_ops tonka_mcbsp_ops = {
 	.hw_params = tonka_mcbsp_hw_params,
 };

 static int tonka_mcbsp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 			struct snd_pcm_hw_params *params)
 {
 	struct snd_interval *channels = hw_param_interval(params,
                                        SNDRV_PCM_HW_PARAM_CHANNELS);

 	channels->min = 2;
 	channels->max = 2;
 	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
 	                            SNDRV_PCM_HW_PARAM_FIRST_MASK],
 	                            SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
 }

 static int sdp4430_stream_event(struct snd_soc_dapm_context *dapm, int event)
 {
 #if 0
 	/*
 	 * set DL1 gains dynamically according to the active output
 	 * (Headset, Earpiece) and HSDAC power mode
 	 */
 	return sdp4430_set_pdm_dl1_gains(dapm);
 #else
 	return 0;
 #endif
 }

 static const struct snd_kcontrol_new tonka_controls_display[] = {
 };

 static const struct snd_soc_dapm_widget tonka_dapm_widgets_display[] = {
 };

 static const struct snd_soc_dapm_route tonka_dapm_routes_display[] = {
 };

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

 	//snd_soc_dapm_nc_pin(dapm, "MIC3L");
 	//…

 	ret = snd_soc_add_controls(codec, tonka_controls_display,
 				   ARRAY_SIZE(tonka_controls_display));
 	if (unlikely(ret < 0))
 		return ret;

 	ret = snd_soc_dapm_new_controls(dapm, tonka_dapm_widgets_display,
 					ARRAY_SIZE(tonka_dapm_widgets_display));
 	if (unlikely(ret < 0))
 		return ret;

 	ret = snd_soc_dapm_add_routes(dapm, tonka_dapm_routes_display,
 				      ARRAY_SIZE(tonka_dapm_routes_display));
 	if (unlikely(ret < 0))
 		return ret;

 	return 0;
 }

 static const struct snd_kcontrol_new tonka_controls_speaker[] = {
 };

 static const struct snd_soc_dapm_widget tonka_dapm_widgets_speaker[] = {
 	SND_SOC_DAPM_LINE("Line Out", NULL),
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_LINE("Line In", NULL),
 	SND_SOC_DAPM_MIC("Mic In", NULL),
 };

 static const struct snd_soc_dapm_route tonka_dapm_routes_speaker[] = {
 	{"Line Out", NULL, "Speaker LOUT"},
 	{"Line Out", NULL, "Speaker ROUT"},
 	{"Headphone Jack", NULL, "Speaker LHPOUT"},
 	{"Headphone Jack", NULL, "Speaker RHPOUT"},
 	{"Speaker LLINEIN", NULL, "Line In"},
 	{"Speaker RLINEIN", NULL, "Line In"},
 	{"Speaker MICIN", NULL, "Mic In"},
 };

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

 	//snd_soc_dapm_nc_pin(dapm, "MIC3L");
 	//…

 	ret = snd_soc_add_controls(codec, tonka_controls_speaker,
 				   ARRAY_SIZE(tonka_controls_speaker));
 	if (unlikely(ret < 0))
 		return ret;

 	ret = snd_soc_dapm_new_controls(dapm, tonka_dapm_widgets_speaker,
 					ARRAY_SIZE(tonka_dapm_widgets_speaker));
 	if (unlikely(ret < 0))
 		return ret;

 	ret = snd_soc_dapm_add_routes(dapm, tonka_dapm_routes_speaker,
 				      ARRAY_SIZE(tonka_dapm_routes_speaker));
 	if (unlikely(ret < 0))
 		return ret;

 	return 0;
 }

 static struct snd_soc_dsp_link fe_media = {
 	.playback	= true,
 	.capture	= true,
 	.trigger =
 		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
 };

 static struct snd_soc_dsp_link fe_media_capture = {
 	.capture	= true,
 	.trigger =
 		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
 };

 static struct snd_soc_dsp_link fe_tones = {
 	.playback	= true,
 	.trigger =
 		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
 };

 static struct snd_soc_dsp_link fe_lp_media = {
 	.playback	= true,
 	.trigger =
 		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
 };

 static struct snd_soc_dai_link tonka_dai[] = {

 /*
  * Frontend DAIs - i.e. userspace visible interfaces (ALSA PCMs)
  */

 	{
 		.name = "Tonka Media",
 		.stream_name = "Multimedia",

 		/* ABE components - MM-UL & MM_DL */
 		.cpu_dai_name = "MultiMedia1",
 		.platform_name = "omap-pcm-audio",

 		.dynamic = 1, /* BE is dynamic */
 		.dsp_link = &fe_media,
 	},
 	{
 		.name = "Tonka Media Capture",
 		.stream_name = "Multimedia Capture",

 		/* ABE components - MM-UL2 */
 		.cpu_dai_name = "MultiMedia2",
 		.platform_name = "omap-pcm-audio",

 		.dynamic = 1, /* BE is dynamic */
 		.dsp_link = &fe_media_capture,
 	},
 	{
 		.name = "Tonka Voice",
 		.stream_name = "Voice",

 		/* ABE components - VX-UL & VX-DL */
 		.cpu_dai_name = "Voice",
 		.platform_name = "omap-pcm-audio",

 		.dynamic = 1, /* BE is dynamic */
 		.dsp_link = &fe_media,
 		.no_host_mode = SND_SOC_DAI_LINK_OPT_HOST,
 	},
 	{
 		.name = "Tonka Tones Playback",
 		.stream_name = "Tone Playback",

 		/* ABE components - TONES_DL */
 		.cpu_dai_name = "Tones",
 		.platform_name = "omap-pcm-audio",

 		.dynamic = 1, /* BE is dynamic */
 		.dsp_link = &fe_tones,
 	},
 	{
 		.name = "Tonka Media LP",
 		.stream_name = "Multimedia",

 		/* ABE components - MM-DL (mmap) */
 		.cpu_dai_name = "MultiMedia1 LP",
 		.platform_name = "aess",

 		.dynamic = 1, /* BE is dynamic */
 		.dsp_link = &fe_lp_media,
 	},

 /*
  * Backend DAIs - i.e. dynamically matched interfaces, invisible to userspace.
  * Matched to above interfaces at runtime, based upon use case.
  */

 	{
 		.name = OMAP_ABE_BE_BT_VX_UL,
 		.stream_name = "BT Capture",

 		/* ABE components - MCBSP2 - BT-VX */
 		.cpu_dai_name = "omap-mcbsp-dai.1",
 		.platform_name = "aess",

 		/* Bluetooth */
 		.codec_dai_name = "tlv320aic23-hifi",
 		.codec_name = "tlv320aic23-codec.2-001b",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBS_CFS,
 		.init = tonka_display_init,

 		.no_pcm = 1, /* don't create ALSA pcm for this */
 		.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
 		.ops = &tonka_mcbsp_ops,
 		.be_id = OMAP_ABE_DAI_BT_VX,
 	},
 	{
 		.name = OMAP_ABE_BE_BT_VX_DL,
 		.stream_name = "BT Playback",

 		/* ABE components - MCBSP2 - BT-VX */
 		.cpu_dai_name = "omap-mcbsp-dai.1",
 		.platform_name = "aess",

 		/* Bluetooth */
 		.codec_dai_name = "tlv320aic23-hifi",
 		.codec_name = "tlv320aic23-codec.2-001b",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBS_CFS,
 		//.init = tonka_display_init,

 		.no_pcm = 1, /* don't create ALSA pcm for this */
 		.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
 		.ops = &tonka_mcbsp_ops,
 		.be_id = OMAP_ABE_DAI_BT_VX,
 	},
 	{
 		.name = "MM-EXT-UL",
 		.stream_name = "FM Capture",

 		/* ABE components - MCBSP1 - MM-EXT */
 		.cpu_dai_name = "omap-mcbsp-dai.0",
 		.platform_name = "aess",

 		/* FM */
 		.codec_dai_name = "tlv320aic23-hifi",
 		.codec_name = "tlv320aic23-codec.3-001a",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBS_CFS,
 		.init = tonka_speaker_init,

 		.no_pcm = 1, /* don't create ALSA pcm for this */
 		.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
 		.ops = &tonka_mcbsp_ops,
 		.be_id = OMAP_ABE_DAI_MM_FM,
 	},
 	{
 		.name = "MM-EXT-DL",
 		.stream_name = "FM Playback",

 		/* ABE components - MCBSP1 - MM-EXT */
 		.cpu_dai_name = "omap-mcbsp-dai.0",
 		.platform_name = "aess",

 		/* FM */
 		.codec_dai_name = "tlv320aic23-hifi",
 		.codec_name = "tlv320aic23-codec.3-001a",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBS_CFS,
 		//.init = tonka_speaker_init,

 		.no_pcm = 1, /* don't create ALSA pcm for this */
 		.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
 		.ops = &tonka_mcbsp_ops,
 		.be_id = OMAP_ABE_DAI_MM_FM,
 	},
 };

 static struct snd_soc_codec_conf tonka_codec_conf[] = {
 	{
 		.dev_name = "tlv320aic23-codec.2-001b",
 		.name_prefix = "Display",
 	},
 	{
 		.dev_name = "tlv320aic23-codec.3-001a",
 		.name_prefix = "Speaker",
 	},
 };

 /* Audio machine driver */
 static struct snd_soc_card snd_soc_tonka = {
 	.name = "Tonka",
 	.driver_name = "OMAP4",
 	.long_name = "TI OMAP4 Board",
 	.dai_link = tonka_dai,
 	.num_links = ARRAY_SIZE(tonka_dai),
 	.codec_conf = tonka_codec_conf,
 	.num_configs = ARRAY_SIZE(tonka_codec_conf),

 	.stream_event = sdp4430_stream_event,
 };

 static struct platform_device *tonka_snd_device;

 static int __init tonka_soc_init(void)
 {
 	int ret;

 	printk(KERN_INFO "Tonka SoC init\n");

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

 	platform_set_drvdata(tonka_snd_device, &snd_soc_tonka);

 	ret = platform_device_add(tonka_snd_device);
 	if (ret)
 		goto err;

 	return 0;

 err:
 	platform_device_put(tonka_snd_device);
 	return ret;
 }
 module_init(tonka_soc_init);

 static void __exit tonka_soc_exit(void)
 {
 	platform_device_unregister(tonka_snd_device);
 }
 module_exit(tonka_soc_exit);

 MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
 MODULE_DESCRIPTION("ALSA SoC Tonka");
 MODULE_LICENSE("GPL");
	/*
	* tonka.c -- SoC audio for Tonka
	*
	* Author: Misael Lopez Cruz <misael.lopez@ti.com>
	* Liam Girdwood <lrg@ti.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.
	*
	* 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., 51 Franklin St, Fifth Floor, Boston, MA
	* 02110-1301 USA
	*
	*/

	#include <linux/clk.h>
	#include <linux/platform_device.h>
	#include <linux/module.h>

	#include <sound/core.h>
	#include <sound/pcm.h>
	#include <sound/soc.h>
	#include <sound/jack.h>
	#include <sound/pcm_params.h>
	#include <sound/soc-dapm.h>
	#include <sound/soc-dsp.h>

	#include <asm/mach-types.h>
	#include <plat/hardware.h>
	#include <plat/mux.h>
	#include <plat/mcbsp.h>

	#include "omap-pcm.h"
	#include "omap-abe.h"
	#include "omap-abe-dsp.h"
	#include "omap-mcbsp.h"

	static int tonka_mcbsp_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 ret = 0;
	unsigned int channels, div;

	/* Codec reference clock. */
	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN);
	if (unlikely(ret < 0)) {
	dev_err(codec_dai->dev, "can't set reference clock\n");
	return ret;
	}

	if (params != NULL) {
	/* Configure McBSP internal buffer usage */
	/* this need to be done for playback and/or record */
	channels = params_channels(params);
	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
	omap_mcbsp_set_tx_threshold(cpu_dai->id, channels);
	else
	omap_mcbsp_set_rx_threshold(cpu_dai->id, channels);
	}

	/* Clock master mode; derive bit clock from FCLK. */
	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK,
	24576000, SND_SOC_CLOCK_IN);
	if (unlikely(ret < 0)) {
	dev_err(cpu_dai->dev, "can't set reference clock\n");
	return ret;
	}

	switch (params_rate(params)) {
	case 44100: /* XXX 45176 */
	div = 17;
	break;
	case 48000:
	div = 16;
	break;
	default:
	dev_err(cpu_dai->dev, "unknown rate %d\n", params_rate(params));
	return -EINVAL;
	}
	ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, div);
	if (unlikely(ret < 0)) {
	dev_err(cpu_dai->dev, "can't set clock divider\n");
	return ret;
	}

	return 0;
	}

	static struct snd_soc_ops tonka_mcbsp_ops = {
	.hw_params = tonka_mcbsp_hw_params,
	};

	static int tonka_mcbsp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
	struct snd_pcm_hw_params *params)
	{
	struct snd_interval *channels = hw_param_interval(params,
	SNDRV_PCM_HW_PARAM_CHANNELS);

	channels->min = 2;
	channels->max = 2;
	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
	SNDRV_PCM_HW_PARAM_FIRST_MASK],
	SNDRV_PCM_FORMAT_S16_LE);
	return 0;
	}

	static int sdp4430_stream_event(struct snd_soc_dapm_context *dapm, int event)
	{
	#if 0
	/*
	* set DL1 gains dynamically according to the active output
	* (Headset, Earpiece) and HSDAC power mode
	*/
	return sdp4430_set_pdm_dl1_gains(dapm);
	#else
	return 0;
	#endif
	}

	static const struct snd_kcontrol_new tonka_controls_display[] = {
	};

	static const struct snd_soc_dapm_widget tonka_dapm_widgets_display[] = {
	};

	static const struct snd_soc_dapm_route tonka_dapm_routes_display[] = {
	};

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

	//snd_soc_dapm_nc_pin(dapm, "MIC3L");
	//…

	ret = snd_soc_add_controls(codec, tonka_controls_display,
	ARRAY_SIZE(tonka_controls_display));
	if (unlikely(ret < 0))
	return ret;

	ret = snd_soc_dapm_new_controls(dapm, tonka_dapm_widgets_display,
	ARRAY_SIZE(tonka_dapm_widgets_display));
	if (unlikely(ret < 0))
	return ret;

	ret = snd_soc_dapm_add_routes(dapm, tonka_dapm_routes_display,
	ARRAY_SIZE(tonka_dapm_routes_display));
	if (unlikely(ret < 0))
	return ret;

	return 0;
	}

	static const struct snd_kcontrol_new tonka_controls_speaker[] = {
	};

	static const struct snd_soc_dapm_widget tonka_dapm_widgets_speaker[] = {
	SND_SOC_DAPM_LINE("Line Out", NULL),
	SND_SOC_DAPM_HP("Headphone Jack", NULL),
	SND_SOC_DAPM_LINE("Line In", NULL),
	SND_SOC_DAPM_MIC("Mic In", NULL),
	};

	static const struct snd_soc_dapm_route tonka_dapm_routes_speaker[] = {
	{"Line Out", NULL, "Speaker LOUT"},
	{"Line Out", NULL, "Speaker ROUT"},
	{"Headphone Jack", NULL, "Speaker LHPOUT"},
	{"Headphone Jack", NULL, "Speaker RHPOUT"},
	{"Speaker LLINEIN", NULL, "Line In"},
	{"Speaker RLINEIN", NULL, "Line In"},
	{"Speaker MICIN", NULL, "Mic In"},
	};

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

	//snd_soc_dapm_nc_pin(dapm, "MIC3L");
	//…

	ret = snd_soc_add_controls(codec, tonka_controls_speaker,
	ARRAY_SIZE(tonka_controls_speaker));
	if (unlikely(ret < 0))
	return ret;

	ret = snd_soc_dapm_new_controls(dapm, tonka_dapm_widgets_speaker,
	ARRAY_SIZE(tonka_dapm_widgets_speaker));
	if (unlikely(ret < 0))
	return ret;

	ret = snd_soc_dapm_add_routes(dapm, tonka_dapm_routes_speaker,
	ARRAY_SIZE(tonka_dapm_routes_speaker));
	if (unlikely(ret < 0))
	return ret;

	return 0;
	}

	static struct snd_soc_dsp_link fe_media = {
	.playback = true,
	.capture = true,
	.trigger =
	{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
	};

	static struct snd_soc_dsp_link fe_media_capture = {
	.capture = true,
	.trigger =
	{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
	};

	static struct snd_soc_dsp_link fe_tones = {
	.playback = true,
	.trigger =
	{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
	};

	static struct snd_soc_dsp_link fe_lp_media = {
	.playback = true,
	.trigger =
	{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
	};

	static struct snd_soc_dai_link tonka_dai[] = {

	/*
	* Frontend DAIs - i.e. userspace visible interfaces (ALSA PCMs)
	*/

	{
	.name = "Tonka Media",
	.stream_name = "Multimedia",

	/* ABE components - MM-UL & MM_DL */
	.cpu_dai_name = "MultiMedia1",
	.platform_name = "omap-pcm-audio",

	.dynamic = 1, /* BE is dynamic */
	.dsp_link = &fe_media,
	},
	{
	.name = "Tonka Media Capture",
	.stream_name = "Multimedia Capture",

	/* ABE components - MM-UL2 */
	.cpu_dai_name = "MultiMedia2",
	.platform_name = "omap-pcm-audio",

	.dynamic = 1, /* BE is dynamic */
	.dsp_link = &fe_media_capture,
	},
	{
	.name = "Tonka Voice",
	.stream_name = "Voice",

	/* ABE components - VX-UL & VX-DL */
	.cpu_dai_name = "Voice",
	.platform_name = "omap-pcm-audio",

	.dynamic = 1, /* BE is dynamic */
	.dsp_link = &fe_media,
	.no_host_mode = SND_SOC_DAI_LINK_OPT_HOST,
	},
	{
	.name = "Tonka Tones Playback",
	.stream_name = "Tone Playback",

	/* ABE components - TONES_DL */
	.cpu_dai_name = "Tones",
	.platform_name = "omap-pcm-audio",

	.dynamic = 1, /* BE is dynamic */
	.dsp_link = &fe_tones,
	},
	{
	.name = "Tonka Media LP",
	.stream_name = "Multimedia",

	/* ABE components - MM-DL (mmap) */
	.cpu_dai_name = "MultiMedia1 LP",
	.platform_name = "aess",

	.dynamic = 1, /* BE is dynamic */
	.dsp_link = &fe_lp_media,
	},

	/*
	* Backend DAIs - i.e. dynamically matched interfaces, invisible to userspace.
	* Matched to above interfaces at runtime, based upon use case.
	*/

	{
	.name = OMAP_ABE_BE_BT_VX_UL,
	.stream_name = "BT Capture",

	/* ABE components - MCBSP2 - BT-VX */
	.cpu_dai_name = "omap-mcbsp-dai.1",
	.platform_name = "aess",

	/* Bluetooth */
	.codec_dai_name = "tlv320aic23-hifi",
	.codec_name = "tlv320aic23-codec.2-001b",
	.dai_fmt = SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \|
	SND_SOC_DAIFMT_CBS_CFS,
	.init = tonka_display_init,

	.no_pcm = 1, /* don't create ALSA pcm for this */
	.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
	.ops = &tonka_mcbsp_ops,
	.be_id = OMAP_ABE_DAI_BT_VX,
	},
	{
	.name = OMAP_ABE_BE_BT_VX_DL,
	.stream_name = "BT Playback",

	/* ABE components - MCBSP2 - BT-VX */
	.cpu_dai_name = "omap-mcbsp-dai.1",
	.platform_name = "aess",

	/* Bluetooth */
	.codec_dai_name = "tlv320aic23-hifi",
	.codec_name = "tlv320aic23-codec.2-001b",
	.dai_fmt = SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \|
	SND_SOC_DAIFMT_CBS_CFS,
	//.init = tonka_display_init,

	.no_pcm = 1, /* don't create ALSA pcm for this */
	.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
	.ops = &tonka_mcbsp_ops,
	.be_id = OMAP_ABE_DAI_BT_VX,
	},
	{
	.name = "MM-EXT-UL",
	.stream_name = "FM Capture",

	/* ABE components - MCBSP1 - MM-EXT */
	.cpu_dai_name = "omap-mcbsp-dai.0",
	.platform_name = "aess",

	/* FM */
	.codec_dai_name = "tlv320aic23-hifi",
	.codec_name = "tlv320aic23-codec.3-001a",
	.dai_fmt = SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \|
	SND_SOC_DAIFMT_CBS_CFS,
	.init = tonka_speaker_init,

	.no_pcm = 1, /* don't create ALSA pcm for this */
	.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
	.ops = &tonka_mcbsp_ops,
	.be_id = OMAP_ABE_DAI_MM_FM,
	},
	{
	.name = "MM-EXT-DL",
	.stream_name = "FM Playback",

	/* ABE components - MCBSP1 - MM-EXT */
	.cpu_dai_name = "omap-mcbsp-dai.0",
	.platform_name = "aess",

	/* FM */
	.codec_dai_name = "tlv320aic23-hifi",
	.codec_name = "tlv320aic23-codec.3-001a",
	.dai_fmt = SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \|
	SND_SOC_DAIFMT_CBS_CFS,
	//.init = tonka_speaker_init,

	.no_pcm = 1, /* don't create ALSA pcm for this */
	.be_hw_params_fixup = tonka_mcbsp_be_hw_params_fixup,
	.ops = &tonka_mcbsp_ops,
	.be_id = OMAP_ABE_DAI_MM_FM,
	},
	};

	static struct snd_soc_codec_conf tonka_codec_conf[] = {
	{
	.dev_name = "tlv320aic23-codec.2-001b",
	.name_prefix = "Display",
	},
	{
	.dev_name = "tlv320aic23-codec.3-001a",
	.name_prefix = "Speaker",
	},
	};

	/* Audio machine driver */
	static struct snd_soc_card snd_soc_tonka = {
	.name = "Tonka",
	.driver_name = "OMAP4",
	.long_name = "TI OMAP4 Board",
	.dai_link = tonka_dai,
	.num_links = ARRAY_SIZE(tonka_dai),
	.codec_conf = tonka_codec_conf,
	.num_configs = ARRAY_SIZE(tonka_codec_conf),

	.stream_event = sdp4430_stream_event,
	};

	static struct platform_device *tonka_snd_device;

	static int __init tonka_soc_init(void)
	{
	int ret;

	printk(KERN_INFO "Tonka SoC init\n");

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

	platform_set_drvdata(tonka_snd_device, &snd_soc_tonka);

	ret = platform_device_add(tonka_snd_device);
	if (ret)
	goto err;

	return 0;

	err:
	platform_device_put(tonka_snd_device);
	return ret;
	}
	module_init(tonka_soc_init);

	static void __exit tonka_soc_exit(void)
	{
	platform_device_unregister(tonka_snd_device);
	}
	module_exit(tonka_soc_exit);

	MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
	MODULE_DESCRIPTION("ALSA SoC Tonka");
	MODULE_LICENSE("GPL");