1 files changed, 349 insertions, 0 deletions
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
new file mode 100644
index 000000000000..93bb8eee22b3
--- /dev/null
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -0,0 +1,349 @@
+/*
+ * omap-abe-twl6040.c  --  SoC audio for TI OMAP based boards with ABE and
+ *                         twl6040 codec
+ *
+ * Author: Misael Lopez Cruz <misael.lopez@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/mfd/twl6040.h>
+#include <linux/platform_data/omap-abe-twl6040.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+#include "omap-dmic.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+#include "../codecs/twl6040.h"
+static int omap_abe_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_codec *codec = rtd->codec;
+        struct snd_soc_card *card = codec->card;
+        struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+        int clk_id, freq;
+        int ret;
+        clk_id = twl6040_get_clk_id(rtd->codec);
+        if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
+                freq = pdata->mclk_freq;
+        else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
+                freq = 32768;
+        else
+                return -EINVAL;
+        /* set the codec mclk */
+        ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
+                                SND_SOC_CLOCK_IN);
+        if (ret) {
+                printk(KERN_ERR "can't set codec system clock\n");
+                return ret;
+        }
+        return ret;
+}
+static struct snd_soc_ops omap_abe_ops = {
+        .hw_params = omap_abe_hw_params,
+};
+static int omap_abe_dmic_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->cpu_dai;
+        int ret = 0;
+        ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
+                                     19200000, SND_SOC_CLOCK_IN);
+        if (ret < 0) {
+                printk(KERN_ERR "can't set DMIC cpu system clock\n");
+                return ret;
+        }
+        ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
+                                     SND_SOC_CLOCK_OUT);
+        if (ret < 0) {
+                printk(KERN_ERR "can't set DMIC output clock\n");
+                return ret;
+        }
+        return 0;
+}
+static struct snd_soc_ops omap_abe_dmic_ops = {
+        .hw_params = omap_abe_dmic_hw_params,
+};
+/* 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,
+        },
+};
+/* SDP4430 machine DAPM */
+static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
+        /* Outputs */
+        SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+        SND_SOC_DAPM_SPK("Earphone Spk", NULL),
+        SND_SOC_DAPM_SPK("Ext Spk", NULL),
+        SND_SOC_DAPM_LINE("Line Out", NULL),
+        SND_SOC_DAPM_SPK("Vibrator", NULL),
+        /* Inputs */
+        SND_SOC_DAPM_MIC("Headset Mic", NULL),
+        SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
+        SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
+        SND_SOC_DAPM_LINE("Line In", NULL),
+};
+static const struct snd_soc_dapm_route audio_map[] = {
+        /* Routings for outputs */
+        {"Headset Stereophone", NULL, "HSOL"},
+        {"Headset Stereophone", NULL, "HSOR"},
+        {"Earphone Spk", NULL, "EP"},
+        {"Ext Spk", NULL, "HFL"},
+        {"Ext Spk", NULL, "HFR"},
+        {"Line Out", NULL, "AUXL"},
+        {"Line Out", NULL, "AUXR"},
+        {"Vibrator", NULL, "VIBRAL"},
+        {"Vibrator", NULL, "VIBRAR"},
+        /* Routings for inputs */
+        {"HSMIC", NULL, "Headset Mic"},
+        {"Headset Mic", NULL, "Headset Mic Bias"},
+        {"MAINMIC", NULL, "Main Handset Mic"},
+        {"Main Handset Mic", NULL, "Main Mic Bias"},
+        {"SUBMIC", NULL, "Sub Handset Mic"},
+        {"Sub Handset Mic", NULL, "Main Mic Bias"},
+        {"AFML", NULL, "Line In"},
+        {"AFMR", NULL, "Line In"},
+};
+static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
+                                          int connected, char *pin)
+{
+        if (!connected)
+                snd_soc_dapm_disable_pin(dapm, pin);
+}
+static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
+{
+        struct snd_soc_codec *codec = rtd->codec;
+        struct snd_soc_card *card = codec->card;
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
+        struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+        int hs_trim;
+        int ret = 0;
+        /* Disable not connected paths if not used */
+        twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
+        twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
+        twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
+        twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
+        twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
+        twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
+        twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
+        twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
+        twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
+        /*
+         * Configure McPDM offset cancellation based on the HSOTRIM value from
+         * twl6040.
+         */
+        hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
+        omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
+                                        TWL6040_HSF_TRIM_RIGHT(hs_trim));
+        /* Headset jack detection only if it is supported */
+        if (pdata->jack_detection) {
+                ret = snd_soc_jack_new(codec, "Headset Jack",
+                                        SND_JACK_HEADSET, &hs_jack);
+                if (ret)
+                        return ret;
+                ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+                                        hs_jack_pins);
+                twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
+        }
+        return ret;
+}
+static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
+        SND_SOC_DAPM_MIC("Digital Mic", NULL),
+};
+static const struct snd_soc_dapm_route dmic_audio_map[] = {
+        {"DMic", NULL, "Digital Mic"},
+        {"Digital Mic", NULL, "Digital Mic1 Bias"},
+};
+static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+        struct snd_soc_codec *codec = rtd->codec;
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
+        int ret;
+        ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
+                                ARRAY_SIZE(dmic_dapm_widgets));
+        if (ret)
+                return ret;
+        return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
+                                ARRAY_SIZE(dmic_audio_map));
+}
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link twl6040_dmic_dai[] = {
+        {
+                .name = "TWL6040",
+                .stream_name = "TWL6040",
+                .cpu_dai_name = "omap-mcpdm",
+                .codec_dai_name = "twl6040-legacy",
+                .platform_name = "omap-pcm-audio",
+                .codec_name = "twl6040-codec",
+                .init = omap_abe_twl6040_init,
+                .ops = &omap_abe_ops,
+        },
+        {
+                .name = "DMIC",
+                .stream_name = "DMIC Capture",
+                .cpu_dai_name = "omap-dmic",
+                .codec_dai_name = "dmic-hifi",
+                .platform_name = "omap-pcm-audio",
+                .codec_name = "dmic-codec",
+                .init = omap_abe_dmic_init,
+                .ops = &omap_abe_dmic_ops,
+        },
+};
+static struct snd_soc_dai_link twl6040_only_dai[] = {
+        {
+                .name = "TWL6040",
+                .stream_name = "TWL6040",
+                .cpu_dai_name = "omap-mcpdm",
+                .codec_dai_name = "twl6040-legacy",
+                .platform_name = "omap-pcm-audio",
+                .codec_name = "twl6040-codec",
+                .init = omap_abe_twl6040_init,
+                .ops = &omap_abe_ops,
+        },
+};
+/* Audio machine driver */
+static struct snd_soc_card omap_abe_card = {
+        .owner = THIS_MODULE,
+        .dapm_widgets = twl6040_dapm_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
+        .dapm_routes = audio_map,
+        .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+static __devinit int omap_abe_probe(struct platform_device *pdev)
+{
+        struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
+        struct snd_soc_card *card = &omap_abe_card;
+        int ret;
+        card->dev = &pdev->dev;
+        if (!pdata) {
+                dev_err(&pdev->dev, "Missing pdata\n");
+                return -ENODEV;
+        }
+        if (pdata->card_name) {
+                card->name = pdata->card_name;
+        } else {
+                dev_err(&pdev->dev, "Card name is not provided\n");
+                return -ENODEV;
+        }
+        if (!pdata->mclk_freq) {
+                dev_err(&pdev->dev, "MCLK frequency missing\n");
+                return -ENODEV;
+        }
+        if (pdata->has_dmic) {
+                card->dai_link = twl6040_dmic_dai;
+                card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
+        } else {
+                card->dai_link = twl6040_only_dai;
+                card->num_links = ARRAY_SIZE(twl6040_only_dai);
+        }
+        ret = snd_soc_register_card(card);
+        if (ret)
+                dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+                        ret);
+        return ret;
+}
+static int __devexit omap_abe_remove(struct platform_device *pdev)
+{
+        struct snd_soc_card *card = platform_get_drvdata(pdev);
+        snd_soc_unregister_card(card);
+        return 0;
+}
+static struct platform_driver omap_abe_driver = {
+        .driver = {
+                .name = "omap-abe-twl6040",
+                .owner = THIS_MODULE,
+                .pm = &snd_soc_pm_ops,
+        },
+        .probe = omap_abe_probe,
+        .remove = __devexit_p(omap_abe_remove),
+};
+module_platform_driver(omap_abe_driver);
+MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-abe-twl6040");

diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c new file mode 100644 index 000000000000..93bb8eee22b3 --- /dev/null +++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -0,0 +1,349 @@
	1	/*
	2	* omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and
	3	* twl6040 codec
	4	*
	5	* Author: Misael Lopez Cruz <misael.lopez@ti.com>
	6	*
	7	* This program is free software; you can redistribute it and/or
	8	* modify it under the terms of the GNU General Public License
	9	* version 2 as published by the Free Software Foundation.
	10	*
	11	* This program is distributed in the hope that it will be useful, but
	12	* WITHOUT ANY WARRANTY; without even the implied warranty of
	13	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	14	* General Public License for more details.
	15	*
	16	* You should have received a copy of the GNU General Public License
	17	* along with this program; if not, write to the Free Software
	18	* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
	19	* 02110-1301 USA
	20	*
	21	*/
	22
	23	#include <linux/clk.h>
	24	#include <linux/platform_device.h>
	25	#include <linux/mfd/twl6040.h>
	26	#include <linux/platform_data/omap-abe-twl6040.h>
	27	#include <linux/module.h>
	28
	29	#include <sound/core.h>
	30	#include <sound/pcm.h>
	31	#include <sound/soc.h>
	32	#include <sound/jack.h>
	33
	34	#include <asm/mach-types.h>
	35	#include <plat/hardware.h>
	36	#include <plat/mux.h>
	37
	38	#include "omap-dmic.h"
	39	#include "omap-mcpdm.h"
	40	#include "omap-pcm.h"
	41	#include "../codecs/twl6040.h"
	42
	43	static int omap_abe_hw_params(struct snd_pcm_substream *substream,
	44	struct snd_pcm_hw_params *params)
	45	{
	46	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	47	struct snd_soc_dai *codec_dai = rtd->codec_dai;
	48	struct snd_soc_codec *codec = rtd->codec;
	49	struct snd_soc_card *card = codec->card;
	50	struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
	51	int clk_id, freq;
	52	int ret;
	53
	54	clk_id = twl6040_get_clk_id(rtd->codec);
	55	if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
	56	freq = pdata->mclk_freq;
	57	else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
	58	freq = 32768;
	59	else
	60	return -EINVAL;
	61
	62	/* set the codec mclk */
	63	ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
	64	SND_SOC_CLOCK_IN);
	65	if (ret) {
	66	printk(KERN_ERR "can't set codec system clock\n");
	67	return ret;
	68	}
	69	return ret;
	70	}
	71
	72	static struct snd_soc_ops omap_abe_ops = {
	73	.hw_params = omap_abe_hw_params,
	74	};
	75
	76	static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream,
	77	struct snd_pcm_hw_params *params)
	78	{
	79	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	80	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
	81	int ret = 0;
	82
	83	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
	84	19200000, SND_SOC_CLOCK_IN);
	85	if (ret < 0) {
	86	printk(KERN_ERR "can't set DMIC cpu system clock\n");
	87	return ret;
	88	}
	89	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
	90	SND_SOC_CLOCK_OUT);
	91	if (ret < 0) {
	92	printk(KERN_ERR "can't set DMIC output clock\n");
	93	return ret;
	94	}
	95	return 0;
	96	}
	97
	98	static struct snd_soc_ops omap_abe_dmic_ops = {
	99	.hw_params = omap_abe_dmic_hw_params,
	100	};
	101
	102	/* Headset jack */
	103	static struct snd_soc_jack hs_jack;
	104
	105	/Headset jack detection DAPM pins /
	106	static struct snd_soc_jack_pin hs_jack_pins[] = {
	107	{
	108	.pin = "Headset Mic",
	109	.mask = SND_JACK_MICROPHONE,
	110	},
	111	{
	112	.pin = "Headset Stereophone",
	113	.mask = SND_JACK_HEADPHONE,
	114	},
	115	};
	116
	117	/* SDP4430 machine DAPM */
	118	static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
	119	/* Outputs */
	120	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
	121	SND_SOC_DAPM_SPK("Earphone Spk", NULL),
	122	SND_SOC_DAPM_SPK("Ext Spk", NULL),
	123	SND_SOC_DAPM_LINE("Line Out", NULL),
	124	SND_SOC_DAPM_SPK("Vibrator", NULL),
	125
	126	/* Inputs */
	127	SND_SOC_DAPM_MIC("Headset Mic", NULL),
	128	SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
	129	SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
	130	SND_SOC_DAPM_LINE("Line In", NULL),
	131	};
	132
	133	static const struct snd_soc_dapm_route audio_map[] = {
	134	/* Routings for outputs */
	135	{"Headset Stereophone", NULL, "HSOL"},
	136	{"Headset Stereophone", NULL, "HSOR"},
	137
	138	{"Earphone Spk", NULL, "EP"},
	139
	140	{"Ext Spk", NULL, "HFL"},
	141	{"Ext Spk", NULL, "HFR"},
	142
	143	{"Line Out", NULL, "AUXL"},
	144	{"Line Out", NULL, "AUXR"},
	145
	146	{"Vibrator", NULL, "VIBRAL"},
	147	{"Vibrator", NULL, "VIBRAR"},
	148
	149	/* Routings for inputs */
	150	{"HSMIC", NULL, "Headset Mic"},
	151	{"Headset Mic", NULL, "Headset Mic Bias"},
	152
	153	{"MAINMIC", NULL, "Main Handset Mic"},
	154	{"Main Handset Mic", NULL, "Main Mic Bias"},
	155
	156	{"SUBMIC", NULL, "Sub Handset Mic"},
	157	{"Sub Handset Mic", NULL, "Main Mic Bias"},
	158
	159	{"AFML", NULL, "Line In"},
	160	{"AFMR", NULL, "Line In"},
	161	};
	162
	163	static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
	164	int connected, char *pin)
	165	{
	166	if (!connected)
	167	snd_soc_dapm_disable_pin(dapm, pin);
	168	}
	169
	170	static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
	171	{
	172	struct snd_soc_codec *codec = rtd->codec;
	173	struct snd_soc_card *card = codec->card;
	174	struct snd_soc_dapm_context *dapm = &codec->dapm;
	175	struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
	176	int hs_trim;
	177	int ret = 0;
	178
	179	/* Disable not connected paths if not used */
	180	twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
	181	twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
	182	twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
	183	twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
	184	twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
	185	twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
	186	twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
	187	twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
	188	twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
	189
	190	/*
	191	* Configure McPDM offset cancellation based on the HSOTRIM value from
	192	* twl6040.
	193	*/
	194	hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
	195	omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
	196	TWL6040_HSF_TRIM_RIGHT(hs_trim));
	197
	198	/* Headset jack detection only if it is supported */
	199	if (pdata->jack_detection) {
	200	ret = snd_soc_jack_new(codec, "Headset Jack",
	201	SND_JACK_HEADSET, &hs_jack);
	202	if (ret)
	203	return ret;
	204
	205	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
	206	hs_jack_pins);
	207	twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
	208	}
	209
	210	return ret;
	211	}
	212
	213	static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
	214	SND_SOC_DAPM_MIC("Digital Mic", NULL),
	215	};
	216
	217	static const struct snd_soc_dapm_route dmic_audio_map[] = {
	218	{"DMic", NULL, "Digital Mic"},
	219	{"Digital Mic", NULL, "Digital Mic1 Bias"},
	220	};
	221
	222	static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
	223	{
	224	struct snd_soc_codec *codec = rtd->codec;
	225	struct snd_soc_dapm_context *dapm = &codec->dapm;
	226	int ret;
	227
	228	ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
	229	ARRAY_SIZE(dmic_dapm_widgets));
	230	if (ret)
	231	return ret;
	232
	233	return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
	234	ARRAY_SIZE(dmic_audio_map));
	235	}
	236
	237	/* Digital audio interface glue - connects codec <--> CPU */
	238	static struct snd_soc_dai_link twl6040_dmic_dai[] = {
	239	{
	240	.name = "TWL6040",
	241	.stream_name = "TWL6040",
	242	.cpu_dai_name = "omap-mcpdm",
	243	.codec_dai_name = "twl6040-legacy",
	244	.platform_name = "omap-pcm-audio",
	245	.codec_name = "twl6040-codec",
	246	.init = omap_abe_twl6040_init,
	247	.ops = &omap_abe_ops,
	248	},
	249	{
	250	.name = "DMIC",
	251	.stream_name = "DMIC Capture",
	252	.cpu_dai_name = "omap-dmic",
	253	.codec_dai_name = "dmic-hifi",
	254	.platform_name = "omap-pcm-audio",
	255	.codec_name = "dmic-codec",
	256	.init = omap_abe_dmic_init,
	257	.ops = &omap_abe_dmic_ops,
	258	},
	259	};
	260
	261	static struct snd_soc_dai_link twl6040_only_dai[] = {
	262	{
	263	.name = "TWL6040",
	264	.stream_name = "TWL6040",
	265	.cpu_dai_name = "omap-mcpdm",
	266	.codec_dai_name = "twl6040-legacy",
	267	.platform_name = "omap-pcm-audio",
	268	.codec_name = "twl6040-codec",
	269	.init = omap_abe_twl6040_init,
	270	.ops = &omap_abe_ops,
	271	},
	272	};
	273
	274	/* Audio machine driver */
	275	static struct snd_soc_card omap_abe_card = {
	276	.owner = THIS_MODULE,
	277
	278	.dapm_widgets = twl6040_dapm_widgets,
	279	.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
	280	.dapm_routes = audio_map,
	281	.num_dapm_routes = ARRAY_SIZE(audio_map),
	282	};
	283
	284	static __devinit int omap_abe_probe(struct platform_device *pdev)
	285	{
	286	struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
	287	struct snd_soc_card *card = &omap_abe_card;
	288	int ret;
	289
	290	card->dev = &pdev->dev;
	291
	292	if (!pdata) {
	293	dev_err(&pdev->dev, "Missing pdata\n");
	294	return -ENODEV;
	295	}
	296
	297	if (pdata->card_name) {
	298	card->name = pdata->card_name;
	299	} else {
	300	dev_err(&pdev->dev, "Card name is not provided\n");
	301	return -ENODEV;
	302	}
	303
	304	if (!pdata->mclk_freq) {
	305	dev_err(&pdev->dev, "MCLK frequency missing\n");
	306	return -ENODEV;
	307	}
	308
	309	if (pdata->has_dmic) {
	310	card->dai_link = twl6040_dmic_dai;
	311	card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
	312	} else {
	313	card->dai_link = twl6040_only_dai;
	314	card->num_links = ARRAY_SIZE(twl6040_only_dai);
	315	}
	316
	317	ret = snd_soc_register_card(card);
	318	if (ret)
	319	dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
	320	ret);
	321
	322	return ret;
	323	}
	324
	325	static int __devexit omap_abe_remove(struct platform_device *pdev)
	326	{
	327	struct snd_soc_card *card = platform_get_drvdata(pdev);
	328
	329	snd_soc_unregister_card(card);
	330
	331	return 0;
	332	}
	333
	334	static struct platform_driver omap_abe_driver = {
	335	.driver = {
	336	.name = "omap-abe-twl6040",
	337	.owner = THIS_MODULE,
	338	.pm = &snd_soc_pm_ops,
	339	},
	340	.probe = omap_abe_probe,
	341	.remove = __devexit_p(omap_abe_remove),
	342	};
	343
	344	module_platform_driver(omap_abe_driver);
	345
	346	MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
	347	MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
	348	MODULE_LICENSE("GPL");
	349	MODULE_ALIAS("platform:omap-abe-twl6040");