Merge remote-tracking branch 'asoc/topic/intel' into asoc-next

8 files changed, 637 insertions, 75 deletions

diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 5639f10774e6..56896e09445d 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -10,6 +10,8 @@ snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
 snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
 snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o
+snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
+snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
 snd-soc-skl_rt286-objs := skl_rt286.o
 snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
 snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
@@ -26,6 +28,8 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 53c6b4cbb1e1..14d9693c1641 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -193,13 +193,12 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd)
 			RT5677_CLK_SEL_I2S1_ASRC);
 
 	/* Request rt5677 GPIO for headphone amp control */
-	bdw_rt5677->gpio_hp_en = devm_gpiod_get_index(codec->dev,
-			"headphone-enable", 0, 0);
+	bdw_rt5677->gpio_hp_en = devm_gpiod_get(codec->dev, "headphone-enable",
+						GPIOD_OUT_LOW);
 	if (IS_ERR(bdw_rt5677->gpio_hp_en)) {
 		dev_err(codec->dev, "Can't find HP_AMP_SHDN_L gpio\n");
 		return PTR_ERR(bdw_rt5677->gpio_hp_en);
 	}
-	gpiod_direction_output(bdw_rt5677->gpio_hp_en, 0);
 
 	/* Create and initialize headphone jack */
 	if (!snd_soc_card_jack_new(rtd->card, "Headphone Jack",
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index faf865bb1765..6dcbbcefc25b 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -269,9 +269,6 @@ static struct snd_soc_card broadwell_rt286 = {
 static int broadwell_audio_probe(struct platform_device *pdev)
 {
 	broadwell_rt286.dev = &pdev->dev;
-
-	snd_soc_set_dmi_name(&broadwell_rt286, NULL);
-
 	return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286);
 }
 
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 2cda06cde4d1..3a8c4d954a91 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -55,6 +55,54 @@ enum {
 	BXT_DPCM_AUDIO_HDMI3_PB,
 };
 
+static inline struct snd_soc_dai *bxt_get_codec_dai(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *rtd;
+
+	list_for_each_entry(rtd, &card->rtd_list, list) {
+
+		if (!strncmp(rtd->codec_dai->name, BXT_DIALOG_CODEC_DAI,
+			     strlen(BXT_DIALOG_CODEC_DAI)))
+			return rtd->codec_dai;
+	}
+
+	return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int  event)
+{
+	int ret = 0;
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+
+	codec_dai = bxt_get_codec_dai(card);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+		return -EIO;
+	}
+
+	if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0,
+			DA7219_SYSCLK_MCLK, 0, 0);
+		if (ret)
+			dev_err(card->dev, "failed to stop PLL: %d\n", ret);
+	} else if(SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = snd_soc_dai_set_sysclk(codec_dai,
+                        DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN);
+		if (ret)
+			dev_err(card->dev, "can't set codec sysclk configuration\n");
+
+		ret = snd_soc_dai_set_pll(codec_dai, 0,
+			DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304);
+		if (ret)
+			dev_err(card->dev, "failed to start PLL: %d\n", ret);
+	}
+
+	return ret;
+}
+
 static const struct snd_kcontrol_new broxton_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
@@ -69,6 +117,8 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = {
 	SND_SOC_DAPM_SPK("HDMI1", NULL),
 	SND_SOC_DAPM_SPK("HDMI2", NULL),
 	SND_SOC_DAPM_SPK("HDMI3", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control,	SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU),
 };
 
 static const struct snd_soc_dapm_route broxton_map[] = {
@@ -109,6 +159,9 @@ static const struct snd_soc_dapm_route broxton_map[] = {
 	/* DMIC */
 	{"dmic01_hifi", NULL, "DMIC01 Rx"},
 	{"DMIC01 Rx", NULL, "DMIC AIF"},
+
+	{ "Headphone Jack", NULL, "Platform Clock" },
+	{ "Headset Mic", NULL, "Platform Clock" },
 };
 
 static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -243,49 +296,6 @@ static const struct snd_soc_ops broxton_da7219_fe_ops = {
 	.startup = bxt_fe_startup,
 };
 
-static int broxton_da7219_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 ret;
-
-	ret = snd_soc_dai_set_sysclk(codec_dai,
-			DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
-
-	ret = snd_soc_dai_set_pll(codec_dai, 0,
-			DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304);
-	if (ret < 0) {
-		dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
-		return -EIO;
-	}
-
-	return ret;
-}
-
-static int broxton_da7219_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int ret;
-
-	ret = snd_soc_dai_set_pll(codec_dai, 0,
-			DA7219_SYSCLK_MCLK, 0, 0);
-	if (ret < 0) {
-		dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
-		return -EIO;
-	}
-
-	return ret;
-}
-
-static const struct snd_soc_ops broxton_da7219_ops = {
-	.hw_params = broxton_da7219_hw_params,
-	.hw_free = broxton_da7219_hw_free,
-};
-
 static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 			struct snd_pcm_hw_params *params)
 {
@@ -467,7 +477,6 @@ static struct snd_soc_dai_link broxton_dais[] = {
 			SND_SOC_DAIFMT_CBS_CFS,
 		.ignore_pmdown_time = 1,
 		.be_hw_params_fixup = broxton_ssp_fixup,
-		.ops = &broxton_da7219_ops,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 	},
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index 176c080a9818..1a68d043c803 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -274,12 +274,15 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream)
 	 * on this platform for PCM device we support:
 	 *      48Khz
 	 *      stereo
+	 *	16-bit audio
 	 */
 
 	runtime->hw.channels_max = 2;
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				&constraints_channels);
 
+	runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
 	snd_pcm_hw_constraint_list(runtime, 0,
 				SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
 
diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c
new file mode 100644
index 000000000000..18873e23f404
--- /dev/null
+++ b/sound/soc/intel/boards/bytcht_da7213.c
@@ -0,0 +1,283 @@
+/*
+ *  bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and
+ *             Cherrytrail-based platforms, with Dialog DA7213 codec
+ *
+ *  Copyright (C) 2017 Intel Corporation
+ *  Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  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/acpi.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <asm/platform_sst_audio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/da7213.h"
+#include "../atom/sst-atom-controls.h"
+#include "../common/sst-acpi.h"
+
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Mic"),
+	SOC_DAPM_PIN_SWITCH("Aux In"),
+};
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPL"},
+	{"Headphone Jack", NULL, "HPR"},
+
+	{"AUXL", NULL, "Aux In"},
+	{"AUXR", NULL, "Aux In"},
+
+	/* Assume Mic1 is linked to Headset and Mic2 to on-board mic */
+	{"MIC1", NULL, "Headset Mic"},
+	{"MIC2", NULL, "Mic"},
+
+	/* SOC-codec link */
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx"},
+	{"codec_in1", NULL, "ssp2 Rx"},
+
+	{"Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Rx", NULL, "Capture"},
+};
+
+static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
+		       struct snd_pcm_hw_params *params)
+{
+	int ret;
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will convert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+
+	/*
+	 * Default mode for SSP configuration is TDM 4 slot, override config
+	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
+	 * dai_set_tdm_slot() since there is no other API exposed
+	 */
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+				  SND_SOC_DAIFMT_I2S     |
+				  SND_SOC_DAIFMT_NB_NF   |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, 48000);
+}
+
+static int aif1_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 ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
+				     19200000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0,
+			DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
+		return -EIO;
+	}
+
+	return ret;
+}
+
+static int aif1_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0,
+				  DA7213_SYSCLK_MCLK, 0, 0);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
+		return -EIO;
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_ops aif1_ops = {
+	.startup = aif1_startup,
+};
+
+static const struct snd_soc_ops ssp2_ops = {
+	.hw_params = aif1_hw_params,
+	.hw_free = aif1_hw_free,
+
+};
+
+static struct snd_soc_dai_link dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &aif1_ops,
+	},
+	[MERR_DPCM_DEEP_BUFFER] = {
+		.name = "Deep-Buffer Audio Port",
+		.stream_name = "Deep-Buffer Audio",
+		.cpu_dai_name = "deepbuffer-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.ops = &aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Compressed Port",
+		.stream_name = "Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "da7213-hifi",
+		.codec_name = "i2c-DLGS7213:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = codec_fixup,
+		.nonatomic = true,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card bytcht_da7213_card = {
+	.name = "bytcht-da7213",
+	.owner = THIS_MODULE,
+	.dai_link = dailink,
+	.num_links = ARRAY_SIZE(dailink),
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
+	.dapm_widgets = dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */
+
+static int bytcht_da7213_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	int i;
+	struct snd_soc_card *card;
+	struct sst_acpi_mach *mach;
+	const char *i2c_name = NULL;
+	int dai_index = 0;
+
+	mach = (&pdev->dev)->platform_data;
+	card = &bytcht_da7213_card;
+	card->dev = &pdev->dev;
+
+	/* fix index of codec dai */
+	dai_index = MERR_DPCM_COMPR + 1;
+	for (i = 0; i < ARRAY_SIZE(dailink); i++) {
+		if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) {
+			dai_index = i;
+			break;
+		}
+	}
+
+	/* fixup codec name based on HID */
+	i2c_name = sst_acpi_find_name_from_hid(mach->id);
+	if (i2c_name != NULL) {
+		snprintf(codec_name, sizeof(codec_name),
+			"%s%s", "i2c-", i2c_name);
+		dailink[dai_index].codec_name = codec_name;
+	}
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, card);
+	return ret_val;
+}
+
+static struct platform_driver bytcht_da7213_driver = {
+	.driver = {
+		.name = "bytcht_da7213",
+	},
+	.probe = bytcht_da7213_probe,
+};
+module_platform_driver(bytcht_da7213_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver");
+MODULE_AUTHOR("Pierre-Louis Bossart");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytcht_da7213");
diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c
new file mode 100644
index 000000000000..89853eeaaf9d
--- /dev/null
+++ b/sound/soc/intel/boards/bytcht_nocodec.c
@@ -0,0 +1,208 @@
+/*
+ *  bytcht_nocodec.c - ASoc Machine driver for MinnowBoard Max and Up
+ *  to make I2S signals observable on the Low-Speed connector. Audio codec
+ *  is not managed by ASoC/DAPM
+ *
+ *  Copyright (C) 2015-2017 Intel Corp
+ *
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  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/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../atom/sst-atom-controls.h"
+
+static const struct snd_soc_dapm_widget widgets[] = {
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("Mic"),
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx"},
+	{"codec_in1", NULL, "ssp2 Rx"},
+
+	{"ssp2 Rx", NULL, "Mic"},
+	{"Speaker", NULL, "ssp2 Tx"},
+};
+
+static int codec_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);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+	int ret;
+
+	/* The DSP will convert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+
+	/*
+	 * Default mode for SSP configuration is TDM 4 slot, override config
+	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
+	 * dai_set_tdm_slot() since there is no other API exposed
+	 */
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+				  SND_SOC_DAIFMT_I2S     |
+				  SND_SOC_DAIFMT_NB_NF   |
+				  SND_SOC_DAIFMT_CBS_CFS);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static unsigned int rates_48000[] = {
+	48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static struct snd_soc_ops aif1_ops = {
+	.startup = aif1_startup,
+};
+
+static struct snd_soc_dai_link dais[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.ignore_suspend = 1,
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &aif1_ops,
+	},
+	[MERR_DPCM_DEEP_BUFFER] = {
+		.name = "Deep-Buffer Audio Port",
+		.stream_name = "Deep-Buffer Audio",
+		.cpu_dai_name = "deepbuffer-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.ignore_suspend = 1,
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.ops = &aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Compressed Port",
+		.stream_name = "Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-LowSpeed Connector",
+		.id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = codec_fixup,
+		.ignore_suspend = 1,
+		.nonatomic = true,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card bytcht_nocodec_card = {
+	.name = "bytcht-nocodec",
+	.owner = THIS_MODULE,
+	.dai_link = dais,
+	.num_links = ARRAY_SIZE(dais),
+	.dapm_widgets = widgets,
+	.num_dapm_widgets = ARRAY_SIZE(widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
+	.fully_routed = true,
+};
+
+static int snd_bytcht_nocodec_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+
+	/* register the soc card */
+	bytcht_nocodec_card.dev = &pdev->dev;
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card);
+
+	if (ret_val) {
+		dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n",
+			ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &bytcht_nocodec_card);
+	return ret_val;
+}
+
+static struct platform_driver snd_bytcht_nocodec_mc_driver = {
+	.driver = {
+		.name = "bytcht_nocodec",
+	},
+	.probe = snd_bytcht_nocodec_mc_probe,
+};
+module_platform_driver(snd_bytcht_nocodec_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Nocodec Machine driver");
+MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytcht_nocodec");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 9e2a3404a836..4a76b099a508 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -19,6 +19,7 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
+#include <linux/moduleparam.h>
 #include <linux/platform_device.h>
 #include <linux/acpi.h>
 #include <linux/device.h>
@@ -56,35 +57,88 @@ enum {
 struct byt_rt5640_private {
 	struct clk *mclk;
 };
+static bool is_bytcr;
 
 static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN;
+static unsigned int quirk_override;
+module_param_named(quirk, quirk_override, uint, 0444);
+MODULE_PARM_DESC(quirk, "Board-specific quirk override");
 
 static void log_quirks(struct device *dev)
 {
-	if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC1_MAP)
-		dev_info(dev, "quirk DMIC1_MAP enabled");
-	if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC2_MAP)
-		dev_info(dev, "quirk DMIC2_MAP enabled");
-	if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN1_MAP)
-		dev_info(dev, "quirk IN1_MAP enabled");
-	if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN3_MAP)
-		dev_info(dev, "quirk IN3_MAP enabled");
-	if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN)
-		dev_info(dev, "quirk DMIC enabled");
+	int map;
+	bool has_dmic = false;
+	bool has_mclk = false;
+	bool has_ssp0 = false;
+	bool has_ssp0_aif1 = false;
+	bool has_ssp0_aif2 = false;
+	bool has_ssp2_aif2 = false;
+
+	map = BYT_RT5640_MAP(byt_rt5640_quirk);
+	switch (map) {
+	case BYT_RT5640_DMIC1_MAP:
+		dev_info(dev, "quirk DMIC1_MAP enabled\n");
+		has_dmic = true;
+		break;
+	case BYT_RT5640_DMIC2_MAP:
+		dev_info(dev, "quirk DMIC2_MAP enabled\n");
+		has_dmic = true;
+		break;
+	case BYT_RT5640_IN1_MAP:
+		dev_info(dev, "quirk IN1_MAP enabled\n");
+		break;
+	case BYT_RT5640_IN3_MAP:
+		dev_info(dev, "quirk IN3_MAP enabled\n");
+		break;
+	default:
+		dev_err(dev, "quirk map 0x%x is not supported, microphone input will not work\n", map);
+		break;
+	}
+	if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) {
+		if (has_dmic)
+			dev_info(dev, "quirk DMIC enabled\n");
+		else
+			dev_err(dev, "quirk DMIC enabled but no DMIC input set, will be ignored\n");
+	}
 	if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER)
-		dev_info(dev, "quirk MONO_SPEAKER enabled");
-	if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC)
-		dev_info(dev, "quirk DIFF_MIC enabled");
-	if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2)
-		dev_info(dev, "quirk SSP2_AIF2 enabled");
-	if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1)
-		dev_info(dev, "quirk SSP0_AIF1 enabled");
-	if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)
-		dev_info(dev, "quirk SSP0_AIF2 enabled");
-	if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN)
-		dev_info(dev, "quirk MCLK_EN enabled");
-	if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ)
-		dev_info(dev, "quirk MCLK_25MHZ enabled");
+		dev_info(dev, "quirk MONO_SPEAKER enabled\n");
+	if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) {
+		if (!has_dmic)
+			dev_info(dev, "quirk DIFF_MIC enabled\n");
+		else
+			dev_info(dev, "quirk DIFF_MIC enabled but DMIC input selected, will be ignored\n");
+	}
+	if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) {
+		dev_info(dev, "quirk SSP0_AIF1 enabled\n");
+		has_ssp0 = true;
+		has_ssp0_aif1 = true;
+	}
+	if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2) {
+		dev_info(dev, "quirk SSP0_AIF2 enabled\n");
+		has_ssp0 = true;
+		has_ssp0_aif2 = true;
+	}
+	if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) {
+		dev_info(dev, "quirk SSP2_AIF2 enabled\n");
+		has_ssp2_aif2 = true;
+	}
+	if (is_bytcr && !has_ssp0)
+		dev_err(dev, "Invalid routing, bytcr detected but no SSP0-based quirk, audio cannot work with SSP2 on bytcr\n");
+	if (has_ssp0_aif1 && has_ssp0_aif2)
+		dev_err(dev, "Invalid routing, SSP0 cannot be connected to both AIF1 and AIF2\n");
+	if (has_ssp0 && has_ssp2_aif2)
+		dev_err(dev, "Invalid routing, cannot have both SSP0 and SSP2 connected to codec\n");
+
+	if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) {
+		dev_info(dev, "quirk MCLK_EN enabled\n");
+		has_mclk = true;
+	}
+	if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) {
+		if (has_mclk)
+			dev_info(dev, "quirk MCLK_25MHZ enabled\n");
+		else
+			dev_err(dev, "quirk MCLK_25MHZ enabled but quirk MCLK not selected, will be ignored\n");
+	}
 }
 
 
@@ -128,7 +182,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w,
 			ret = clk_prepare_enable(priv->mclk);
 			if (ret < 0) {
 				dev_err(card->dev,
-					"could not configure MCLK state");
+					"could not configure MCLK state\n");
 				return ret;
 			}
 		}
@@ -710,8 +764,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
 	int i;
 	int dai_index;
 	struct byt_rt5640_private *priv;
-	bool is_bytcr = false;
 
+	is_bytcr = false;
 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC);
 	if (!priv)
 		return -ENOMEM;
@@ -806,6 +860,11 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
 
 	/* check quirks before creating card */
 	dmi_check_system(byt_rt5640_quirk_table);
+	if (quirk_override) {
+		dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n",
+			 (unsigned int)byt_rt5640_quirk, quirk_override);
+		byt_rt5640_quirk = quirk_override;
+	}
 	log_quirks(&pdev->dev);
 
 	if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) ||