Minimum files for lt-omap package creation

1 files changed, 1137 insertions, 0 deletions

diff --git a/debian.linaro/ci/rr-cache/f5615ad0da79534d9fd4f6cbd714c4fb37335145/preimage b/debian.linaro/ci/rr-cache/f5615ad0da79534d9fd4f6cbd714c4fb37335145/preimage
new file mode 100644
index 0000000..60d2ad3
--- /dev/null
+++ b/debian.linaro/ci/rr-cache/f5615ad0da79534d9fd4f6cbd714c4fb37335145/preimage
@@ -0,0 +1,1137 @@
+/*
+ * sdp4430.c  --  SoC audio for TI OMAP4430 SDP
+ *
+ * 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/mfd/twl6040.h>
+#include <linux/i2c.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-mcpdm.h"
+#include "omap-pcm.h"
+#include "omap-abe.h"
+#include "omap-abe-dsp.h"
+#include "omap-mcbsp.h"
+#include "omap-dmic.h"
+#include "../codecs/twl6040.h"
+
+static int twl6040_power_mode;
+static int mcbsp_cfg;
+static struct snd_soc_codec *twl6040_codec;
+static struct i2c_client *tps6130x_client;
+static struct i2c_board_info tps6130x_hwmon_info = {
+	I2C_BOARD_INFO("tps6130x", 0x33),
+};
+
+/* configure the TPS6130x Handsfree Boost Converter */
+static int sdp4430_tps6130x_configure(void)
+{
+	u8 data[2];
+
+	data[0] = 0x01;
+	data[1] = 0x60;
+	if (i2c_master_send(tps6130x_client, data, 2) != 2)
+		printk(KERN_ERR "I2C write to TPS6130x failed\n");
+
+	data[0] = 0x02;
+	if (i2c_master_send(tps6130x_client, data, 2) != 2)
+		printk(KERN_ERR "I2C write to TPS6130x failed\n");
+	return 0;
+}
+
+static int sdp4430_modem_mcbsp_configure(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params, int flag)
+{
+	int ret = 0;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_pcm_substream *modem_substream[2];
+	struct snd_soc_pcm_runtime *modem_rtd;
+	int channels;
+
+	if (flag) {
+		modem_substream[substream->stream] =
+		snd_soc_get_dai_substream(rtd->card,
+						OMAP_ABE_BE_MM_EXT1,
+						substream->stream);
+		if (unlikely(modem_substream[substream->stream] == NULL))
+			return -ENODEV;
+
+		modem_rtd =
+			modem_substream[substream->stream]->private_data;
+
+		if (!mcbsp_cfg) {
+			/* Set cpu DAI configuration */
+			ret = snd_soc_dai_set_fmt(modem_rtd->cpu_dai,
+					SND_SOC_DAIFMT_I2S |
+					SND_SOC_DAIFMT_NB_NF |
+					SND_SOC_DAIFMT_CBM_CFM);
+			if (unlikely(ret < 0)) {
+				printk(KERN_ERR "can't set Modem cpu DAI configuration\n");
+				goto exit;
+				} else {
+					mcbsp_cfg = 1;
+			}
+		}
+
+		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_rx_threshold(
+					modem_rtd->cpu_dai->id, channels);
+			else
+				omap_mcbsp_set_tx_threshold(
+					modem_rtd->cpu_dai->id, channels);
+		}
+	} else {
+		mcbsp_cfg = 0;
+	}
+
+exit:
+	return ret;
+}
+
+static int sdp4430_modem_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	int ret;
+
+	ret = sdp4430_modem_mcbsp_configure(substream, params, 1);
+	if (ret)
+		printk(KERN_ERR "can't set modem cpu DAI configuration\n");
+
+	return ret;
+}
+
+static int sdp4430_modem_hw_free(struct snd_pcm_substream *substream)
+{
+	int ret;
+
+	ret = sdp4430_modem_mcbsp_configure(substream, NULL, 0);
+	if (ret)
+		printk(KERN_ERR "can't clear modem cpu DAI configuration\n");
+
+	return ret;
+}
+
+static struct snd_soc_ops sdp4430_modem_ops = {
+	.hw_params = sdp4430_modem_hw_params,
+	.hw_free = sdp4430_modem_hw_free,
+};
+static int sdp4430_mcpdm_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 clk_id, freq, ret;
+
+	if (twl6040_power_mode) {
+		clk_id = TWL6040_SYSCLK_SEL_HPPLL;
+		freq = 38400000;
+	} else {
+		clk_id = TWL6040_SYSCLK_SEL_LPPLL;
+		freq = 32768;
+	}
+
+	/* 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;
+}
+
+static struct snd_soc_ops sdp4430_mcpdm_ops = {
+	.hw_params = sdp4430_mcpdm_hw_params,
+};
+
+static int sdp4430_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 *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+	unsigned int be_id, channels;
+
+        be_id = rtd->dai_link->be_id;
+
+	if (be_id == OMAP_ABE_DAI_BT_VX) {
+		ret = snd_soc_dai_set_fmt(cpu_dai,
+			SND_SOC_DAIFMT_DSP_B |
+				SND_SOC_DAIFMT_NB_IF |
+			SND_SOC_DAIFMT_CBM_CFM);
+	} else {
+		/* Set cpu DAI configuration */
+		ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	}
+
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\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);
+	}
+
+	/*
+	 * TODO: where does this clock come from (external source??) -
+	 * do we need to enable it.
+	 */
+	/* Set McBSP clock to external */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK,
+				     64 * params_rate(params),
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		printk(KERN_ERR "can't set cpu system clock\n");
+
+	return ret;
+}
+
+static struct snd_soc_ops sdp4430_mcbsp_ops = {
+	.hw_params = sdp4430_mcbsp_hw_params,
+};
+
+static int sdp4430_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_clkdiv(cpu_dai, OMAP_DMIC_CLKDIV, 8);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set DMIC cpu clock divider\n");
+		return ret;
+	}
+	return 0;
+}
+
+static struct snd_soc_ops sdp4430_dmic_ops = {
+	.hw_params = sdp4430_dmic_hw_params,
+};
+static int 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);
+	unsigned int be_id = rtd->dai_link->be_id;
+
+	if (be_id == OMAP_ABE_DAI_MM_FM)
+		channels->min = 2;
+	else if (be_id == OMAP_ABE_DAI_BT_VX)
+		channels->min = 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 dmic_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+
+	/* The ABE will covert the FE rate to 96k */
+	rate->min = rate->max = 96000;
+
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+	                            SNDRV_PCM_HW_PARAM_FIRST_MASK],
+	                            SNDRV_PCM_FORMAT_S32_LE);
+	return 0;
+}
+
+/* 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,
+	},
+};
+
+static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = twl6040_power_mode;
+	return 0;
+}
+
+static int sdp4430_set_power_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	if (twl6040_power_mode == ucontrol->value.integer.value[0])
+		return 0;
+
+	twl6040_power_mode = ucontrol->value.integer.value[0];
+	abe_dsp_set_power_mode(twl6040_power_mode);
+
+	return 1;
+}
+
+static const char *power_texts[] = {"Low-Power", "High-Performance"};
+
+static const struct soc_enum sdp4430_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, power_texts),
+};
+
+static const struct snd_kcontrol_new sdp4430_controls[] = {
+	SOC_ENUM_EXT("TWL6040 Power Mode", sdp4430_enum[0],
+		sdp4430_get_power_mode, sdp4430_set_power_mode),
+};
+/* SDP4430 machine DAPM */
+static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_SPK("Earphone Spk", NULL),
+	SND_SOC_DAPM_INPUT("FM Stereo In"),
+
+	SND_SOC_DAPM_MIC("Digital Mic 0", NULL),
+	SND_SOC_DAPM_MIC("Digital Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Digital Mic 2", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Main Mic Bias"},
+	{"SUBMIC", NULL, "Main Mic Bias"},
+	{"Main Mic Bias", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Headset Stereophone (Headphone): HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	/* Earphone speaker */
+	{"Earphone Spk", NULL, "EP"},
+
+	/* Aux/FM Stereo In: AFML, AFMR */
+	{"AFML", NULL, "FM Stereo In"},
+	{"AFMR", NULL, "FM Stereo In"},
+
+	/* Digital Mics: DMic0, DMic1, DMic2 with bias */
+	{"DMIC0", NULL, "Digital Mic1 Bias"},
+	{"Digital Mic1 Bias", NULL, "Digital Mic 0"},
+
+	{"DMIC1", NULL, "Digital Mic1 Bias"},
+	{"Digital Mic1 Bias", NULL, "Digital Mic 1"},
+
+	{"DMIC2", NULL, "Digital Mic1 Bias"},
+	{"Digital Mic1 Bias", NULL, "Digital Mic 2"},
+};
+
+static int sdp4430_set_pdm_dl1_gains(struct snd_soc_dapm_context *dapm)
+{
+	int output, val;
+
+	if (snd_soc_dapm_get_pin_power(dapm, "Earphone Spk")) {
+		output = OMAP_ABE_DL1_EARPIECE;
+	} else if (snd_soc_dapm_get_pin_power(dapm, "Headset Stereophone")) {
+		val = snd_soc_read(twl6040_codec, TWL6040_REG_HSLCTL);
+		if (val & TWL6040_HSDACMODE)
+			/* HSDACL in LP mode */
+			output = OMAP_ABE_DL1_HEADSET_LP;
+		else
+			/* HSDACL in HP mode */
+			output = OMAP_ABE_DL1_HEADSET_HP;
+	} else {
+		output = OMAP_ABE_DL1_NO_PDM;
+	}
+
+	return omap_abe_set_dl1_output(output);
+}
+
+static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct twl6040 *twl6040 = codec->control_data;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int hsotrim, left_offset, right_offset, mode;
+	int ret;
+
+	/* Add SDP4430 specific controls */
+	ret = snd_soc_add_controls(codec, sdp4430_controls,
+				ARRAY_SIZE(sdp4430_controls));
+	if (ret)
+		return ret;
+
+	/* Add SDP4430 specific widgets */
+	ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets,
+				ARRAY_SIZE(sdp4430_twl6040_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up SDP4430 specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	/* SDP4430 connected pins */
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "AFML");
+	snd_soc_dapm_enable_pin(dapm, "AFMR");
+	snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+
+	/* allow audio paths from the audio modem to run during suspend */
+	snd_soc_dapm_ignore_suspend(dapm, "Ext Mic");
+	snd_soc_dapm_ignore_suspend(dapm, "Ext Spk");
+	snd_soc_dapm_ignore_suspend(dapm, "AFML");
+	snd_soc_dapm_ignore_suspend(dapm, "AFMR");
+	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
+	snd_soc_dapm_ignore_suspend(dapm, "Headset Stereophone");
+	snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 0");
+	snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 1");
+	snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 2");
+
+	ret = snd_soc_dapm_sync(dapm);
+	if (ret)
+		return ret;
+
+	/* Headset 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);
+
+	if (machine_is_omap_4430sdp())
+		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
+	else
+		snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET);
+
+	/* DC offset cancellation computation */
+	hsotrim = snd_soc_read(codec, TWL6040_REG_HSOTRIM);
+	right_offset = (hsotrim & TWL6040_HSRO) >> TWL6040_HSRO_OFFSET;
+	left_offset = hsotrim & TWL6040_HSLO;
+
+	if ((twl6040_get_revid(twl6040) == TWL6040_REV_ES1_0) ||
+		(twl6040_get_revid(twl6040) == TWL6040_REV_ES1_1))
+		/* For ES under ES_1.0 and 1.1 HS step is 2 mV */
+		mode = 2;
+	else
+		/* For ES_1.3 HS step is 1 mV */
+		mode = 1;
+
+	abe_dsp_set_hs_offset(left_offset, right_offset, mode);
+
+	/* don't wait before switching of HS power */
+	rtd->pmdown_time = 0;
+
+	return ret;
+}
+
+static int sdp4430_twl6040_dl2_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int hfotrim, left_offset, right_offset;
+
+	/* DC offset cancellation computation */
+	hfotrim = snd_soc_read(codec, TWL6040_REG_HFOTRIM);
+	right_offset = (hfotrim & TWL6040_HFRO) >> TWL6040_HFRO_OFFSET;
+	left_offset = hfotrim & TWL6040_HFLO;
+
+	abe_dsp_set_hf_offset(left_offset, right_offset);
+
+	/* don't wait before switching of HF power */
+	rtd->pmdown_time = 0;
+	return 0;
+}
+
+static int sdp4430_twl6040_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+
+	/* don't wait before switching of FE power */
+	rtd->pmdown_time = 0;
+
+	return 0;
+}
+
+static int sdp4430_bt_init(struct snd_soc_pcm_runtime *rtd)
+{
+
+	/* don't wait before switching of BT power */
+	rtd->pmdown_time = 0;
+
+	return 0;
+}
+
+static int sdp4430_stream_event(struct snd_soc_dapm_context *dapm, int event)
+{
+	/*
+	 * set DL1 gains dynamically according to the active output
+	 * (Headset, Earpiece) and HSDAC power mode
+	 */
+	return sdp4430_set_pdm_dl1_gains(dapm);
+}
+
+/* SDP4430 digital microphones DAPM */
+static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Digital Mic Legacy", NULL),
+};
+
+static const struct snd_soc_dapm_route dmic_audio_map[] = {
+	{"DMic", NULL, "Digital Mic1 Bias"},
+	{"Digital Mic1 Bias", NULL, "Digital Mic Legacy"},
+};
+
+static int sdp4430_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, sdp4430_dmic_dapm_widgets,
+				ARRAY_SIZE(sdp4430_dmic_dapm_widgets));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_add_routes(dapm, dmic_audio_map,
+				ARRAY_SIZE(dmic_audio_map));
+	if (ret)
+		return ret;
+
+	snd_soc_dapm_enable_pin(dapm, "Digital Mic Legacy");
+
+	ret = snd_soc_dapm_sync(dapm);
+
+	return ret;
+}
+
+/* TODO: make this a separate BT CODEC driver or DUMMY */
+static struct snd_soc_dai_driver dai[] = {
+{
+	.name = "Bluetooth",
+	.playback = {
+		.stream_name = "BT Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+					SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "BT Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+					SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+/* TODO: make this a separate FM CODEC driver or DUMMY */
+{
+	.name = "FM Digital",
+	.playback = {
+		.stream_name = "FM Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "FM Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "HDMI",
+	.playback = {
+		.stream_name = "HDMI Playback",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+				SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+};
+
+struct snd_soc_dsp_link fe_media = {
+	.playback	= true,
+	.capture	= true,
+	.trigger =
+		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
+};
+
+struct snd_soc_dsp_link fe_media_capture = {
+	.capture	= true,
+	.trigger =
+		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
+};
+
+struct snd_soc_dsp_link fe_tones = {
+	.playback	= true,
+	.trigger =
+		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
+};
+
+struct snd_soc_dsp_link fe_vib = {
+	.playback	= true,
+	.trigger =
+		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
+};
+
+struct snd_soc_dsp_link fe_modem = {
+	.playback	= true,
+	.capture	= true,
+	.trigger =
+		{SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE},
+};
+
+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 sdp4430_dai[] = {
+
+/*
+ * Frontend DAIs - i.e. userspace visible interfaces (ALSA PCMs)
+ */
+
+	{
+		.name = "SDP4430 Media",
+		.stream_name = "Multimedia",
+
+		/* ABE components - MM-UL & MM_DL */
+		.cpu_dai_name = "MultiMedia1",
+		.platform_name = "omap-pcm-audio",
+
+		.dynamic = 1, /* BE is dynamic */
+		.init = sdp4430_twl6040_fe_init,
+		.dsp_link = &fe_media,
+	},
+	{
+		.name = "SDP4430 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 = "SDP4430 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 = "SDP4430 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 = "SDP4430 Vibra Playback",
+		.stream_name = "VIB-DL",
+
+		/* ABE components - DMIC UL 2 */
+		.cpu_dai_name = "Vibra",
+		.platform_name = "omap-pcm-audio",
+
+		.dynamic = 1, /* BE is dynamic */
+		.dsp_link = &fe_vib,
+	},
+	{
+		.name = "SDP4430 MODEM",
+		.stream_name = "MODEM",
+
+		/* ABE components - MODEM <-> McBSP2 */
+		.cpu_dai_name = "MODEM",
+		.platform_name = "aess",
+
+		.dynamic = 1, /* BE is dynamic */
+		.init = sdp4430_twl6040_fe_init,
+		.dsp_link = &fe_modem,
+		.ops = &sdp4430_modem_ops,
+		.no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = "SDP4430 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,
+	},
+	{
+		.name = "Legacy McBSP",
+		.stream_name = "Multimedia",
+
+		/* ABE components - MCBSP2 - MM-EXT */
+		.cpu_dai_name = "omap-mcbsp-dai.1",
+		.platform_name = "omap-pcm-audio",
+
+		/* FM */
+		.codec_dai_name = "FM Digital",
+
+		.no_codec = 1, /* TODO: have a dummy CODEC */
+		.ops = &sdp4430_mcbsp_ops,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = "Legacy McPDM",
+		.stream_name = "Headset Playback",
+
+		/* ABE components - DL1 */
+		.cpu_dai_name = "mcpdm-dl",
+		.platform_name = "omap-pcm-audio",
+
+		/* Phoenix - DL1 DAC */
+		.codec_dai_name =  "twl6040-legacy",
+		.codec_name = "twl6040-codec",
+
+		.ops = &sdp4430_mcpdm_ops,
+		.ignore_suspend = 1,
+	},
+#if 0
+	{
+		.name = "SPDIF",
+		.stream_name = "SPDIF",
+		.cpu_dai_name = "omap-mcasp-dai.0",
+		.codec_dai_name = "dit-hifi",	/* dummy s/pdif transciever
+						 * driver */
+		.platform_name = "omap-pcm-audio",
+		.ignore_suspend = 1,
+		.no_codec = 1,
+	},
+#endif
+	{
+		.name = "Legacy DMIC",
+		.stream_name = "DMIC Capture",
+
+		/* ABE components - DMIC0 */
+		.cpu_dai_name = "omap-dmic-dai-0",
+		.platform_name = "omap-pcm-audio",
+
+		/* DMIC codec */
+		.codec_dai_name = "dmic-hifi",
+		.codec_name = "dmic-codec.0",
+
+		.init = sdp4430_dmic_init,
+		.ops = &sdp4430_dmic_ops,
+	},
+
+/*
+ * Backend DAIs - i.e. dynamically matched interfaces, invisible to userspace.
+ * Matched to above interfaces at runtime, based upon use case.
+ */
+
+	{
+		.name = OMAP_ABE_BE_PDM_DL1,
+		.stream_name = "HS Playback",
+
+		/* ABE components - DL1 */
+		.cpu_dai_name = "mcpdm-dl1",
+		.platform_name = "aess",
+
+		/* Phoenix - DL1 DAC */
+		.codec_dai_name =  "twl6040-dl1",
+		.codec_name = "twl6040-codec",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.init = sdp4430_twl6040_init,
+		.ops = &sdp4430_mcpdm_ops,
+		.be_id = OMAP_ABE_DAI_PDM_DL1,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = OMAP_ABE_BE_PDM_UL1,
+		.stream_name = "Analog Capture",
+
+		/* ABE components - UL1 */
+		.cpu_dai_name = "mcpdm-ul1",
+		.platform_name = "aess",
+
+		/* Phoenix - UL ADC */
+		.codec_dai_name =  "twl6040-ul",
+		.codec_name = "twl6040-codec",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.ops = &sdp4430_mcpdm_ops,
+		.be_id = OMAP_ABE_DAI_PDM_UL,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = OMAP_ABE_BE_PDM_DL2,
+		.stream_name = "HF Playback",
+
+		/* ABE components - DL2 */
+		.cpu_dai_name = "mcpdm-dl2",
+		.platform_name = "aess",
+
+		/* Phoenix - DL2 DAC */
+		.codec_dai_name =  "twl6040-dl2",
+		.codec_name = "twl6040-codec",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.init = sdp4430_twl6040_dl2_init,
+		.ops = &sdp4430_mcpdm_ops,
+		.be_id = OMAP_ABE_DAI_PDM_DL2,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = OMAP_ABE_BE_PDM_VIB,
+		.stream_name = "Vibra",
+
+		/* ABE components - VIB1 DL */
+		.cpu_dai_name = "mcpdm-vib",
+		.platform_name = "aess",
+
+		/* Phoenix - PDM to PWM */
+		.codec_dai_name =  "twl6040-vib",
+		.codec_name = "twl6040-codec",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.ops = &sdp4430_mcpdm_ops,
+		.be_id = OMAP_ABE_DAI_PDM_VIB,
+	},
+	{
+		.name = OMAP_ABE_BE_BT_VX_UL,
+		.stream_name = "BT Capture",
+
+		/* ABE components - MCBSP1 - BT-VX */
+		.cpu_dai_name = "omap-mcbsp-dai.0",
+		.platform_name = "aess",
+
+		/* Bluetooth */
+		.codec_dai_name = "Bluetooth",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.no_codec = 1, /* TODO: have a dummy CODEC */
+		.be_hw_params_fixup = mcbsp_be_hw_params_fixup,
+		.ops = &sdp4430_mcbsp_ops,
+		.be_id = OMAP_ABE_DAI_BT_VX,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = OMAP_ABE_BE_BT_VX_DL,
+		.stream_name = "BT Playback",
+
+		/* ABE components - MCBSP1 - BT-VX */
+		.cpu_dai_name = "omap-mcbsp-dai.0",
+		.platform_name = "aess",
+
+		/* Bluetooth */
+		.codec_dai_name = "Bluetooth",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.no_codec = 1, /* TODO: have a dummy CODEC */
+		.init = sdp4430_bt_init,
+		.be_hw_params_fixup = mcbsp_be_hw_params_fixup,
+		.ops = &sdp4430_mcbsp_ops,
+		.be_id = OMAP_ABE_DAI_BT_VX,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = OMAP_ABE_BE_MM_EXT0,
+		.stream_name = "FM",
+
+		/* ABE components - MCBSP2 - MM-EXT */
+		.cpu_dai_name = "omap-mcbsp-dai.1",
+		.platform_name = "aess",
+
+		/* FM */
+		.codec_dai_name = "FM Digital",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.no_codec = 1, /* TODO: have a dummy CODEC */
+		.be_hw_params_fixup = mcbsp_be_hw_params_fixup,
+		.ops = &sdp4430_mcbsp_ops,
+		.be_id = OMAP_ABE_DAI_MM_FM,
+	},
+	{
+		.name = OMAP_ABE_BE_MM_EXT1,
+		.stream_name = "MODEM",
+
+		/* ABE components - MCBSP2 - MM-EXT */
+		.cpu_dai_name = "omap-mcbsp-dai.1",
+		.platform_name = "aess",
+
+		/* MODEM */
+		.codec_dai_name = "MODEM",
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.no_codec = 1, /* TODO: have a dummy CODEC */
+		.be_hw_params_fixup = mcbsp_be_hw_params_fixup,
+		.ops = &sdp4430_mcbsp_ops,
+		.be_id = OMAP_ABE_DAI_MODEM,
+		.ignore_suspend = 1,
+	},
+	{
+		.name = OMAP_ABE_BE_DMIC0,
+		.stream_name = "DMIC0 Capture",
+
+		/* ABE components - DMIC UL 1 */
+		.cpu_dai_name = "omap-dmic-abe-dai-0",
+		.platform_name = "aess",
+
+		/* DMIC 0 */
+		.codec_dai_name = "dmic-hifi",
+		.codec_name = "dmic-codec.0",
+		.ops = &sdp4430_dmic_ops,
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.be_hw_params_fixup = dmic_be_hw_params_fixup,
+		.be_id = OMAP_ABE_DAI_DMIC0,
+	},
+	{
+		.name = OMAP_ABE_BE_DMIC1,
+		.stream_name = "DMIC1 Capture",
+
+		/* ABE components - DMIC UL 1 */
+		.cpu_dai_name = "omap-dmic-abe-dai-1",
+		.platform_name = "aess",
+
+		/* DMIC 1 */
+		.codec_dai_name = "dmic-hifi",
+		.codec_name = "dmic-codec.1",
+		.ops = &sdp4430_dmic_ops,
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.be_hw_params_fixup = dmic_be_hw_params_fixup,
+		.be_id = OMAP_ABE_DAI_DMIC1,
+	},
+	{
+		.name = OMAP_ABE_BE_DMIC2,
+		.stream_name = "DMIC2 Capture",
+
+		/* ABE components - DMIC UL 2 */
+		.cpu_dai_name = "omap-dmic-abe-dai-2",
+		.platform_name = "aess",
+
+		/* DMIC 2 */
+		.codec_dai_name = "dmic-hifi",
+		.codec_name = "dmic-codec.2",
+		.ops = &sdp4430_dmic_ops,
+
+		.no_pcm = 1, /* don't create ALSA pcm for this */
+		.be_hw_params_fixup = dmic_be_hw_params_fixup,
+		.be_id = OMAP_ABE_DAI_DMIC2,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_sdp4430 = {
+	.driver_name = "OMAP4",
+	.long_name = "TI OMAP4 Board",
+	.dai_link = sdp4430_dai,
+	.num_links = ARRAY_SIZE(sdp4430_dai),
+
+	.stream_event = sdp4430_stream_event,
+};
+
+<<<<<<<
+static int __devinit sdp4430_soc_probe(struct platform_device *pdev)
+=======
+static struct platform_device *sdp4430_snd_device;
+static struct i2c_adapter *adapter;
+
+static int __init sdp4430_soc_init(void)
+>>>>>>>
+{
+	struct snd_soc_card *card = &snd_soc_sdp4430;
+	int ret;
+
+<<<<<<<
+	if (!machine_is_omap_4430sdp() && !machine_is_omap4_panda()) {
+		pr_debug("Not SDP4430 or PandaBoard!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "SDP4430 SoC init\n");
+	if (machine_is_omap_4430sdp())
+		snd_soc_sdp4430.name = "SDP4430";
+	else if (machine_is_omap4_panda())
+		snd_soc_sdp4430.name = "Panda";
+=======
+	pr_info("SDP4430 SoC init\n");
+
+	card->dev = &pdev->dev;
+>>>>>>>
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		return ret;
+	}
+
+<<<<<<<
+	ret = snd_soc_register_dais(&sdp4430_snd_device->dev, dai, ARRAY_SIZE(dai));
+	if (ret < 0)
+		goto err_dai;
+	platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430);
+
+	ret = platform_device_add(sdp4430_snd_device);
+	if (ret)
+		goto err_dev;
+
+	adapter = i2c_get_adapter(1);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter\n");
+		ret = -ENODEV;
+		goto err_adap;
+	}
+
+	tps6130x_client = i2c_new_device(adapter, &tps6130x_hwmon_info);
+	if (!tps6130x_client) {
+		printk(KERN_ERR "can't add i2c device\n");
+		ret = -ENODEV;
+		goto err_i2c;
+	}
+
+	/* Only configure the TPS6130x on SDP4430 */
+	if (machine_is_omap_4430sdp())
+		sdp4430_tps6130x_configure();
+
+	twl6040_codec = snd_soc_card_get_codec(&snd_soc_sdp4430,
+					"twl6040-codec");
+=======
+	return 0;
+}
+
+static int __devexit sdp4430_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+>>>>>>>
+
+	return 0;
+}
+
+<<<<<<<
+err_i2c:
+	i2c_put_adapter(adapter);
+err_adap:
+	platform_device_del(sdp4430_snd_device);
+err_dev:
+	snd_soc_unregister_dais(&sdp4430_snd_device->dev, ARRAY_SIZE(dai));
+err_dai:
+	platform_device_put(sdp4430_snd_device);
+	return ret;
+=======
+static struct platform_driver sdp4430_driver = {
+	.driver = {
+		.name = "sdp4430-soc-audio",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = sdp4430_soc_probe,
+	.remove = __devexit_p(sdp4430_soc_remove),
+};
+
+static int __init sdp4430_soc_init(void)
+{
+	return platform_driver_register(&sdp4430_driver);
+>>>>>>>
+}
+module_init(sdp4430_soc_init);
+
+static void __exit sdp4430_soc_exit(void)
+{
+<<<<<<<
+	platform_device_unregister(sdp4430_snd_device);
+	snd_soc_unregister_dais(&sdp4430_snd_device->dev, ARRAY_SIZE(dai));
+	i2c_unregister_device(tps6130x_client);
+	i2c_put_adapter(adapter);
+=======
+	platform_driver_unregister(&sdp4430_driver);
+>>>>>>>
+}
+module_exit(sdp4430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP4430");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sdp4430-soc-audio");