Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/ml26124' and...

Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/ml26124' and 'asoc/topic/omap' into asoc-next

Documentation/devicetree/bindings/sound/mt8173-max98090.txt

+MT8173 with MAX98090 CODEC
+
+Required properties:
+- compatible : "mediatek,mt8173-max98090"
+- mediatek,audio-codec: the phandle of the MAX98090 audio codec
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt8173-max98090";
+		mediatek,audio-codec = <&max98090>;
+	};
+

Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt

+MT8173 with RT5650 RT5676 CODECS
+
+Required properties:
+- compatible : "mediatek,mt8173-rt5650-rt5676"
+- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt8173-rt5650-rt5676";
+		mediatek,audio-codec = <&rt5650 &rt5676>;
+	};
+

Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt

+Mediatek AFE PCM controller
+
+Required properties:
+- compatible = "mediatek,mt8173-afe-pcm";
+- reg: register location and size
+- interrupts: Should contain AFE interrupt
+- clock-names: should have these clock names:
+		"infra_sys_audio_clk",
+		"top_pdn_audio",
+		"top_pdn_aud_intbus",
+		"bck0",
+		"bck1",
+		"i2s0_m",
+		"i2s1_m",
+		"i2s2_m",
+		"i2s3_m",
+		"i2s3_b";
+
+Example:
+
+	afe: mt8173-afe-pcm@11220000  {
+		compatible = "mediatek,mt8173-afe-pcm";
+		reg = <0 0x11220000 0 0x1000>;
+		interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>;
+		clocks = <&infracfg INFRA_AUDIO>,
+			<&topckgen TOP_AUDIO_SEL>,
+			<&topckgen TOP_AUD_INTBUS_SEL>,
+			<&topckgen TOP_APLL1_DIV0>,
+			<&topckgen TOP_APLL2_DIV0>,
+			<&topckgen TOP_I2S0_M_CK_SEL>,
+			<&topckgen TOP_I2S1_M_CK_SEL>,
+			<&topckgen TOP_I2S2_M_CK_SEL>,
+			<&topckgen TOP_I2S3_M_CK_SEL>,
+			<&topckgen TOP_I2S3_B_CK_SEL>;
+		clock-names = "infra_sys_audio_clk",
+				"top_pdn_audio",
+				"top_pdn_aud_intbus",
+				"bck0",
+				"bck1",
+				"i2s0_m",
+				"i2s1_m",
+				"i2s2_m",
+				"i2s3_m",
+				"i2s3_b";
+	};

sound/soc/Kconfig

@@ -45,6 +45,7 @@ source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/kirkwood/Kconfig"
 source "sound/soc/intel/Kconfig"
+source "sound/soc/mediatek/Kconfig"
 source "sound/soc/mxs/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/qcom/Kconfig"

sound/soc/Makefile

@@ -24,6 +24,7 @@ obj-$(CONFIG_SND_SOC)	+= dwc/
 obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
 obj-$(CONFIG_SND_SOC)	+= intel/
+obj-$(CONFIG_SND_SOC)	+= mediatek/
 obj-$(CONFIG_SND_SOC)	+= mxs/
 obj-$(CONFIG_SND_SOC)	+= nuc900/
 obj-$(CONFIG_SND_SOC)	+= omap/

sound/soc/codecs/ml26124.c

@@ -341,6 +341,7 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = dai->codec;
 	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int i = get_coeff(priv->mclk, params_rate(hw_params));
+	int srate;
 
 	if (i < 0)
 		return i;
@@ -370,53 +371,16 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream,
 				    BIT(0) | BIT(1), 0);
 	}
 
-	switch (params_rate(hw_params)) {
-	case 16000:
-		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
-				    get_srate(params_rate(hw_params)));
-		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
-				    coeff_div[i].pllnl);
-		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
-				    coeff_div[i].pllnh);
-		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
-				    coeff_div[i].pllml);
-		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
-				    coeff_div[i].pllmh);
-		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
-				    coeff_div[i].plldiv);
-		break;
-	case 32000:
-		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
-				    get_srate(params_rate(hw_params)));
-		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
-				    coeff_div[i].pllnl);
-		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
-				    coeff_div[i].pllnh);
-		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
-				    coeff_div[i].pllml);
-		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
-				    coeff_div[i].pllmh);
-		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
-				    coeff_div[i].plldiv);
-		break;
-	case 48000:
-		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
-				    get_srate(params_rate(hw_params)));
-		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
-				    coeff_div[i].pllnl);
-		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
-				    coeff_div[i].pllnh);
-		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
-				    coeff_div[i].pllml);
-		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
-				    coeff_div[i].pllmh);
-		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
-				    coeff_div[i].plldiv);
-		break;
-	default:
-		pr_err("%s:this rate is no support for ml26124\n", __func__);
-		return -EINVAL;
-	}
+	srate = get_srate(params_rate(hw_params));
+	if (srate < 0)
+		return srate;
+
+	snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, srate);
+	snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, coeff_div[i].pllnl);
+	snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, coeff_div[i].pllnh);
+	snd_soc_update_bits(codec, ML26124_PLLML, 0xff, coeff_div[i].pllml);
+	snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, coeff_div[i].pllmh);
+	snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, coeff_div[i].plldiv);
 
 	return 0;
 }

sound/soc/mediatek/Kconfig

+config SND_SOC_MEDIATEK
+	tristate "ASoC support for Mediatek chip"
+	depends on ARCH_MEDIATEK
+	help
+	  This adds ASoC platform driver support for Mediatek chip
+	  that can be used with other codecs.
+	  Select Y if you have such device.
+	  Ex: MT8173
+
+config SND_SOC_MT8173_MAX98090
+	tristate "ASoC Audio driver for MT8173 with MAX98090 codec"
+	depends on SND_SOC_MEDIATEK
+	select SND_SOC_MAX98090
+	help
+	  This adds ASoC driver for Mediatek MT8173 boards
+	  with the MAX98090 audio codec.
+	  Select Y if you have such device.
+	  If unsure select "N".
+
+config SND_SOC_MT8173_RT5650_RT5676
+	tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs"
+	depends on SND_SOC_MEDIATEK
+	select SND_SOC_RT5645
+	select SND_SOC_RT5677
+	help
+	  This adds ASoC driver for Mediatek MT8173 boards
+	  with the RT5650 and RT5676 codecs.
+	  Select Y if you have such device.
+	  If unsure select "N".
+

sound/soc/mediatek/Makefile

+# MTK Platform Support
+obj-$(CONFIG_SND_SOC_MEDIATEK) += mtk-afe-pcm.o
+# Machine support
+obj-$(CONFIG_SND_SOC_MT8173_MAX98090) += mt8173-max98090.o
+obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o

sound/soc/mediatek/mt8173-max98090.c

+/*
+ * mt8173-max98090.c  --  MT8173 MAX98090 ALSA SoC machine driver
+ *
+ * Copyright (c) 2015 MediaTek Inc.
+ * Author: Koro Chen <koro.chen@mediatek.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 and
+ * only 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.
+ */
+
+#include <linux/module.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <linux/gpio.h>
+#include "../codecs/max98090.h"
+
+static struct snd_soc_jack mt8173_max98090_jack;
+
+static struct snd_soc_jack_pin mt8173_max98090_jack_pins[] = {
+	{
+		.pin	= "Headphone",
+		.mask	= SND_JACK_HEADPHONE,
+	},
+	{
+		.pin	= "Headset Mic",
+		.mask	= SND_JACK_MICROPHONE,
+	},
+};
+
+static const struct snd_soc_dapm_widget mt8173_max98090_widgets[] = {
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mt8173_max98090_routes[] = {
+	{"Speaker", NULL, "SPKL"},
+	{"Speaker", NULL, "SPKR"},
+	{"DMICL", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPL"},
+	{"Headphone", NULL, "HPR"},
+	{"Headset Mic", NULL, "MICBIAS"},
+	{"IN34", NULL, "Headset Mic"},
+};
+
+static const struct snd_kcontrol_new mt8173_max98090_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static int mt8173_max98090_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;
+
+	return snd_soc_dai_set_sysclk(codec_dai, 0, params_rate(params) * 256,
+				      SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops mt8173_max98090_ops = {
+	.hw_params = mt8173_max98090_hw_params,
+};
+
+static int mt8173_max98090_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_card *card = runtime->card;
+	struct snd_soc_codec *codec = runtime->codec;
+
+	/* enable jack detection */
+	ret = snd_soc_card_jack_new(card, "Headphone", SND_JACK_HEADPHONE,
+				    &mt8173_max98090_jack, NULL, 0);
+	if (ret) {
+		dev_err(card->dev, "Can't snd_soc_jack_new %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_jack_add_pins(&mt8173_max98090_jack,
+				    ARRAY_SIZE(mt8173_max98090_jack_pins),
+				    mt8173_max98090_jack_pins);
+	if (ret) {
+		dev_err(card->dev, "Can't snd_soc_jack_add_pins %d\n", ret);
+		return ret;
+	}
+
+	return max98090_mic_detect(codec, &mt8173_max98090_jack);
+}
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link mt8173_max98090_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "MAX98090 Playback",
+		.stream_name = "MAX98090 Playback",
+		.cpu_dai_name = "DL1",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dynamic = 1,
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "MAX98090 Capture",
+		.stream_name = "MAX98090 Capture",
+		.cpu_dai_name = "VUL",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dynamic = 1,
+		.dpcm_capture = 1,
+	},
+	/* Back End DAI links */
+	{
+		.name = "Codec",
+		.cpu_dai_name = "I2S",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.no_pcm = 1,
+		.codec_dai_name = "HiFi",
+		.init = mt8173_max98090_init,
+		.ops = &mt8173_max98090_ops,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+static struct snd_soc_card mt8173_max98090_card = {
+	.name = "mt8173-max98090",
+	.dai_link = mt8173_max98090_dais,
+	.num_links = ARRAY_SIZE(mt8173_max98090_dais),
+	.controls = mt8173_max98090_controls,
+	.num_controls = ARRAY_SIZE(mt8173_max98090_controls),
+	.dapm_widgets = mt8173_max98090_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mt8173_max98090_widgets),
+	.dapm_routes = mt8173_max98090_routes,
+	.num_dapm_routes = ARRAY_SIZE(mt8173_max98090_routes),
+};
+
+static int mt8173_max98090_dev_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &mt8173_max98090_card;
+	struct device_node *codec_node;
+	int ret, i;
+
+	codec_node = of_parse_phandle(pdev->dev.of_node,
+				      "mediatek,audio-codec", 0);
+	if (!codec_node) {
+		dev_err(&pdev->dev,
+			"Property 'audio-codec' missing or invalid\n");
+		return -EINVAL;
+	}
+	for (i = 0; i < card->num_links; i++) {
+		if (mt8173_max98090_dais[i].codec_name)
+			continue;
+		mt8173_max98090_dais[i].codec_of_node = codec_node;
+	}
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
+			__func__, ret);
+	return ret;
+}
+
+static int mt8173_max98090_dev_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static const struct of_device_id mt8173_max98090_dt_match[] = {
+	{ .compatible = "mediatek,mt8173-max98090", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, mt8173_max98090_dt_match);
+
+static struct platform_driver mt8173_max98090_driver = {
+	.driver = {
+		   .name = "mt8173-max98090",
+		   .owner = THIS_MODULE,
+		   .of_match_table = mt8173_max98090_dt_match,
+#ifdef CONFIG_PM
+		   .pm = &snd_soc_pm_ops,
+#endif
+	},
+	.probe = mt8173_max98090_dev_probe,
+	.remove = mt8173_max98090_dev_remove,
+};
+
+module_platform_driver(mt8173_max98090_driver);
+
+/* Module information */
+MODULE_DESCRIPTION("MT8173 MAX98090 ALSA SoC machine driver");
+MODULE_AUTHOR("Koro Chen <koro.chen@mediatek.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:mt8173-max98090");
+

sound/soc/mediatek/mt8173-rt5650-rt5676.c

+/*
+ * mt8173-rt5650-rt5676.c  --  MT8173 machine driver with RT5650/5676 codecs
+ *
+ * Copyright (c) 2015 MediaTek Inc.
+ * Author: Koro Chen <koro.chen@mediatek.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 and
+ * only 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.
+ */
+
+#include <linux/module.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/rt5645.h"
+#include "../codecs/rt5677.h"
+
+#define MCLK_FOR_CODECS		12288000
+
+static const struct snd_soc_dapm_widget mt8173_rt5650_rt5676_widgets[] = {
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mt8173_rt5650_rt5676_routes[] = {
+	{"Speaker", NULL, "SPOL"},
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650  */
+	{"Sub DMIC L1", NULL, "Int Mic"}, /* DMIC from 5676 */
+	{"Sub DMIC R1", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Headphone", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650  */
+	{"Headset Mic", NULL, "micbias1"},
+	{"Headset Mic", NULL, "micbias2"},
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"Sub AIF2RX", NULL, "Headset Mic"}, /* IF2 DAC from 5650  */
+};
+
+static const struct snd_kcontrol_new mt8173_rt5650_rt5676_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream,
+					  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int i, ret;
+
+	for (i = 0; i < rtd->num_codecs; i++) {
+		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+
+		/* pll from mclk 12.288M */
+		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
+					  params_rate(params) * 512);
+		if (ret)
+			return ret;
+
+		/* sysclk from pll */
+		ret = snd_soc_dai_set_sysclk(codec_dai, 1,
+					     params_rate(params) * 512,
+					     SND_SOC_CLOCK_IN);
+		if (ret)
+			return ret;
+	}
+	return 0;
+}
+
+static struct snd_soc_ops mt8173_rt5650_rt5676_ops = {
+	.hw_params = mt8173_rt5650_rt5676_hw_params,
+};
+
+static struct snd_soc_jack mt8173_rt5650_rt5676_jack;
+
+static int mt8173_rt5650_rt5676_init(struct snd_soc_pcm_runtime *runtime)
+{
+	struct snd_soc_card *card = runtime->card;
+	struct snd_soc_codec *codec = runtime->codec_dais[0]->codec;
+	struct snd_soc_codec *codec_sub = runtime->codec_dais[1]->codec;
+	int ret;
+
+	rt5645_sel_asrc_clk_src(codec,
+				RT5645_DA_STEREO_FILTER |
+				RT5645_AD_STEREO_FILTER,
+				RT5645_CLK_SEL_I2S1_ASRC);
+	rt5677_sel_asrc_clk_src(codec_sub,
+				RT5677_DA_STEREO_FILTER |
+				RT5677_AD_STEREO1_FILTER,
+				RT5677_CLK_SEL_I2S1_ASRC);
+	rt5677_sel_asrc_clk_src(codec_sub,
+				RT5677_AD_STEREO2_FILTER |
+				RT5677_I2S2_SOURCE,
+				RT5677_CLK_SEL_I2S2_ASRC);
+
+	/* enable jack detection */
+	ret = snd_soc_card_jack_new(card, "Headset Jack",
+				    SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+				    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+				    SND_JACK_BTN_2 | SND_JACK_BTN_3,
+				    &mt8173_rt5650_rt5676_jack, NULL, 0);
+	if (ret) {
+		dev_err(card->dev, "Can't new Headset Jack %d\n", ret);
+		return ret;
+	}
+
+	return rt5645_set_jack_detect(codec,
+				      &mt8173_rt5650_rt5676_jack,
+				      &mt8173_rt5650_rt5676_jack,
+				      &mt8173_rt5650_rt5676_jack);
+}
+
+static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = {
+	{
+		.dai_name = "rt5645-aif1",
+	},
+	{
+		.dai_name = "rt5677-aif1",
+	},
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "rt5650_rt5676 Playback",
+		.stream_name = "rt5650_rt5676 Playback",
+		.cpu_dai_name = "DL1",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dynamic = 1,
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "rt5650_rt5676 Capture",
+		.stream_name = "rt5650_rt5676 Capture",
+		.cpu_dai_name = "VUL",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dynamic = 1,
+		.dpcm_capture = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		.name = "Codec",
+		.cpu_dai_name = "I2S",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.no_pcm = 1,
+		.codecs = mt8173_rt5650_rt5676_codecs,
+		.num_codecs = 2,
+		.init = mt8173_rt5650_rt5676_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &mt8173_rt5650_rt5676_ops,
+		.ignore_pmdown_time = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{ /* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */
+		.name = "rt5650_rt5676 intercodec",
+		.stream_name = "rt5650_rt5676 intercodec",
+		.cpu_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5677-aif2",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBM_CFM,
+	},
+
+};
+
+static struct snd_soc_codec_conf mt8173_rt5650_rt5676_codec_conf[] = {
+	{
+		.name_prefix = "Sub",
+	},
+};
+
+static struct snd_soc_card mt8173_rt5650_rt5676_card = {
+	.name = "mtk-rt5650-rt5676",
+	.dai_link = mt8173_rt5650_rt5676_dais,
+	.num_links = ARRAY_SIZE(mt8173_rt5650_rt5676_dais),
+	.codec_conf = mt8173_rt5650_rt5676_codec_conf,
+	.num_configs = ARRAY_SIZE(mt8173_rt5650_rt5676_codec_conf),
+	.controls = mt8173_rt5650_rt5676_controls,
+	.num_controls = ARRAY_SIZE(mt8173_rt5650_rt5676_controls),
+	.dapm_widgets = mt8173_rt5650_rt5676_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_rt5676_widgets),
+	.dapm_routes = mt8173_rt5650_rt5676_routes,
+	.num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_rt5676_routes),
+};
+
+static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
+	int ret;
+
+	mt8173_rt5650_rt5676_codecs[0].of_node =
+		of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0);
+	if (!mt8173_rt5650_rt5676_codecs[0].of_node) {
+		dev_err(&pdev->dev,
+			"Property 'audio-codec' missing or invalid\n");
+		return -EINVAL;
+	}
+	mt8173_rt5650_rt5676_codecs[1].of_node =
+		of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1);
+	if (!mt8173_rt5650_rt5676_codecs[1].of_node) {
+		dev_err(&pdev->dev,
+			"Property 'audio-codec' missing or invalid\n");
+		return -EINVAL;
+	}
+	mt8173_rt5650_rt5676_codec_conf[0].of_node =
+		mt8173_rt5650_rt5676_codecs[1].of_node;
+
+	mt8173_rt5650_rt5676_dais[3].codec_of_node =
+		mt8173_rt5650_rt5676_codecs[1].of_node;
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
+			__func__, ret);
+	return ret;
+}
+
+static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = {
+	{ .compatible = "mediatek,mt8173-rt5650-rt5676", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, mt8173_rt5650_rt5676_dt_match);
+
+static struct platform_driver mt8173_rt5650_rt5676_driver = {
+	.driver = {
+		   .name = "mtk-rt5650-rt5676",
+		   .owner = THIS_MODULE,
+		   .of_match_table = mt8173_rt5650_rt5676_dt_match,
+#ifdef CONFIG_PM
+		   .pm = &snd_soc_pm_ops,
+#endif
+	},
+	.probe = mt8173_rt5650_rt5676_dev_probe,
+	.remove = mt8173_rt5650_rt5676_dev_remove,
+};
+
+module_platform_driver(mt8173_rt5650_rt5676_driver);
+
+/* Module information */
+MODULE_DESCRIPTION("MT8173 RT5650 and RT5676 SoC machine driver");
+MODULE_AUTHOR("Koro Chen <koro.chen@mediatek.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:mtk-rt5650-rt5676");
+

sound/soc/mediatek/mtk-afe-common.h

+/*
+ * mtk_afe_common.h  --  Mediatek audio driver common definitions
+ *
+ * Copyright (c) 2015 MediaTek Inc.
+ * Author: Koro Chen <koro.chen@mediatek.com>
+ *             Sascha Hauer <s.hauer@pengutronix.de>
+ *             Hidalgo Huang <hidalgo.huang@mediatek.com>
+ *             Ir Lian <ir.lian@mediatek.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 and
+ * only 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.
+ */
+
+#ifndef _MTK_AFE_COMMON_H_
+#define _MTK_AFE_COMMON_H_
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+
+enum {
+	MTK_AFE_MEMIF_DL1,
+	MTK_AFE_MEMIF_DL2,
+	MTK_AFE_MEMIF_VUL,
+	MTK_AFE_MEMIF_DAI,
+	MTK_AFE_MEMIF_AWB,
+	MTK_AFE_MEMIF_MOD_DAI,
+	MTK_AFE_MEMIF_HDMI,
+	MTK_AFE_MEMIF_NUM,
+	MTK_AFE_IO_MOD_PCM1 = MTK_AFE_MEMIF_NUM,
+	MTK_AFE_IO_MOD_PCM2,
+	MTK_AFE_IO_PMIC,
+	MTK_AFE_IO_I2S,
+	MTK_AFE_IO_2ND_I2S,
+	MTK_AFE_IO_HW_GAIN1,
+	MTK_AFE_IO_HW_GAIN2,
+	MTK_AFE_IO_MRG_O,
+	MTK_AFE_IO_MRG_I,
+	MTK_AFE_IO_DAIBT,
+	MTK_AFE_IO_HDMI,
+};
+
+enum {
+	MTK_AFE_IRQ_1,
+	MTK_AFE_IRQ_2,
+	MTK_AFE_IRQ_3,
+	MTK_AFE_IRQ_4,
+	MTK_AFE_IRQ_5,
+	MTK_AFE_IRQ_6,
+	MTK_AFE_IRQ_7,
+	MTK_AFE_IRQ_8,
+	MTK_AFE_IRQ_NUM,
+};
+
+enum {
+	MTK_CLK_INFRASYS_AUD,
+	MTK_CLK_TOP_PDN_AUD,
+	MTK_CLK_TOP_PDN_AUD_BUS,
+	MTK_CLK_I2S0_M,
+	MTK_CLK_I2S1_M,
+	MTK_CLK_I2S2_M,
+	MTK_CLK_I2S3_M,
+	MTK_CLK_I2S3_B,
+	MTK_CLK_BCK0,
+	MTK_CLK_BCK1,
+	MTK_CLK_NUM
+};
+
+struct mtk_afe;
+struct snd_pcm_substream;
+
+struct mtk_afe_memif_data {
+	int id;
+	const char *name;
+	int reg_ofs_base;
+	int reg_ofs_cur;
+	int fs_shift;
+	int mono_shift;
+	int enable_shift;
+	int irq_reg_cnt;
+	int irq_cnt_shift;
+	int irq_en_shift;
+	int irq_fs_shift;
+	int irq_clr_shift;
+};
+
+struct mtk_afe_memif {
+	unsigned int phys_buf_addr;
+	int buffer_size;
+	unsigned int hw_ptr;		/* Previous IRQ's HW ptr */
+	struct snd_pcm_substream *substream;
+	const struct mtk_afe_memif_data *data;
+	const struct mtk_afe_irq_data *irqdata;
+};
+
+struct mtk_afe {
+	/* address for ioremap audio hardware register */
+	void __iomem *base_addr;
+	struct device *dev;
+	struct regmap *regmap;
+	struct mtk_afe_memif memif[MTK_AFE_MEMIF_NUM];
+	struct clk *clocks[MTK_CLK_NUM];
+};
+#endif

sound/soc/omap/rx51.c

@@ -245,6 +245,8 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
 static const struct snd_soc_dapm_route audio_map[] = {
 	{"Ext Spk", NULL, "HPLOUT"},
 	{"Ext Spk", NULL, "HPROUT"},
+	{"Ext Spk", NULL, "HPLCOM"},
+	{"Ext Spk", NULL, "HPRCOM"},
 	{"Headphone Jack", NULL, "LLOUT"},
 	{"Headphone Jack", NULL, "RLOUT"},
 	{"FM Transmitter", NULL, "LLOUT"},
@@ -288,15 +290,8 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_card *card = rtd->card;
 	struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
-
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
-	/* Set up NC codec pins */
-	snd_soc_dapm_nc_pin(dapm, "MIC3L");
-	snd_soc_dapm_nc_pin(dapm, "MIC3R");
-	snd_soc_dapm_nc_pin(dapm, "LINE1R");
-
 	err = tpa6130a2_add_controls(codec);
 	if (err < 0) {
 		dev_err(card->dev, "Failed to add TPA6130A2 controls\n");
@@ -383,6 +378,7 @@ static struct snd_soc_card rx51_sound_card = {
 	.num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
 	.codec_conf = rx51_codec_conf,
 	.num_configs = ARRAY_SIZE(rx51_codec_conf),
+	.fully_routed = true,
 
 	.controls = aic34_rx51_controls,
 	.num_controls = ARRAY_SIZE(aic34_rx51_controls),