Merge tag 'sound-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There are little changes in core part, but lots of development are
  found in drivers, especially ASoC.  The diffstat shows regmap-related
  changes for a slight API additions / changes, and that's all.

  Looking at the code size statistics, the most significant addition is
  for Intel Skylake.  (Note that SKL support is still underway, the
  codec driver is missing.) Also STI controller driver is a major
  addition as well as a few new codec drivers.

  In HD-audio side, there are fewer changes than the past.  The
  noticeable change is the support of ELD notification from i915
  graphics driver.  Thus this pull request carries a few changes in
  drm/i915.

  Other than that, USB-audio got a rewrite of runtime PM code.  It was
  initiated by lockdep warning, but resulted in a good cleanup in the
  end.

  Below are the highlights:

  Common:
   - Factoring out of AC'97 reset code from ASoC into the core helper
   - A few regmap API extensions (in case it's not pulled yet)

  ASoC:
   - New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
     RT298 and ST STI controllers
   - Machine drivers for Rockchip systems with MAX98090 and RT5645 and
     RT5650
   - Initial driver support for Intel Skylake devices
   - Lots of rsnd cleanup and enhancements
   - A few DAPM fixes and cleanups
   - A large number of cleanups in various drivers (conversion and
     standardized to regmap, component) mostly by Lars-Peter and Axel

  HD-audio:
   - Extended HD-audio core for Intel Skylake controller support
   - Quirks for Dell headsets, Alienware 15
   - Clean up of pin-based quirk tables for Realtek codecs
   - ELD notifier implenetation for Intel HDMI/DP

  USB-audio:
   - Refactor runtime PM code to make lockdep happier"

* tag 'sound-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
  drm/i915: Add locks around audio component bind/unbind
  drm/i915: Drop port_mst_index parameter from pin/eld callback
  ALSA: hda - Fix missing inline for dummy snd_hdac_set_codec_wakeup()
  ALSA: hda - Wake the codec up on pin/ELD notify events
  ALSA: hda - allow codecs to access the i915 pin/ELD callback
  drm/i915: Call audio pin/ELD notify function
  drm/i915: Add audio pin sense / ELD callback
  ASoC: zx296702-i2s: Fix resource leak when unload module
  ASoC: sti_uniperif: Ensure component is unregistered when unload module
  ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource
  ASoC: sh: dma-sh7760: Convert to devm_snd_soc_register_platform
  ASoC: spear_pcm: Use devm_snd_dmaengine_pcm_register to fix resource leak
  ALSA: fireworks/bebob/dice/oxfw: fix substreams counting at vmalloc failure
  ASoC: Clean up docbook warnings
  ASoC: txx9: Convert to devm_snd_soc_register_platform
  ASoC: pxa: Convert to devm_snd_soc_register_platform
  ASoC: nuc900: Convert to devm_snd_soc_register_platform
  ASoC: blackfin: Convert to devm_snd_soc_register_platform
  ASoC: au1x: Convert to devm_snd_soc_register_platform
  ASoC: qcom: Constify asoc_qcom_lpass_cpu_dai_ops
  ...

1 files changed, 628 insertions, 0 deletions

diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
new file mode 100644
index 000000000000..160d61a66204
--- /dev/null
+++ b/sound/soc/codecs/sti-sas.c
@@ -0,0 +1,628 @@
+/*
+ * Copyright (C) STMicroelectronics SA 2015
+ * Authors: Arnaud Pouliquen <arnaud.pouliquen@st.com>
+ *          for STMicroelectronics.
+ * License terms:  GNU General Public License (GPL), version 2
+ */
+
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/reset.h>
+#include <linux/mfd/syscon.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+/* chipID supported */
+#define CHIPID_STIH416 0
+#define CHIPID_STIH407 1
+
+/* DAC definitions */
+
+/* stih416 DAC registers */
+/* sysconf 2517: Audio-DAC-Control */
+#define STIH416_AUDIO_DAC_CTRL 0x00000814
+/* sysconf 2519: Audio-Gue-Control */
+#define STIH416_AUDIO_GLUE_CTRL 0x0000081C
+
+#define STIH416_DAC_NOT_STANDBY	0x3
+#define STIH416_DAC_SOFTMUTE	0x4
+#define STIH416_DAC_ANA_NOT_PWR	0x5
+#define STIH416_DAC_NOT_PNDBG	0x6
+
+#define STIH416_DAC_NOT_STANDBY_MASK	BIT(STIH416_DAC_NOT_STANDBY)
+#define STIH416_DAC_SOFTMUTE_MASK	BIT(STIH416_DAC_SOFTMUTE)
+#define STIH416_DAC_ANA_NOT_PWR_MASK	BIT(STIH416_DAC_ANA_NOT_PWR)
+#define STIH416_DAC_NOT_PNDBG_MASK	BIT(STIH416_DAC_NOT_PNDBG)
+
+/* stih407 DAC registers */
+/* sysconf 5041: Audio-Gue-Control */
+#define STIH407_AUDIO_GLUE_CTRL 0x000000A4
+/* sysconf 5042: Audio-DAC-Control */
+#define STIH407_AUDIO_DAC_CTRL 0x000000A8
+
+/* DAC definitions */
+#define STIH407_DAC_SOFTMUTE		0x0
+#define STIH407_DAC_STANDBY_ANA		0x1
+#define STIH407_DAC_STANDBY		0x2
+
+#define STIH407_DAC_SOFTMUTE_MASK	BIT(STIH407_DAC_SOFTMUTE)
+#define STIH407_DAC_STANDBY_ANA_MASK    BIT(STIH407_DAC_STANDBY_ANA)
+#define STIH407_DAC_STANDBY_MASK        BIT(STIH407_DAC_STANDBY)
+
+/* SPDIF definitions */
+#define SPDIF_BIPHASE_ENABLE		0x6
+#define SPDIF_BIPHASE_IDLE		0x7
+
+#define SPDIF_BIPHASE_ENABLE_MASK	BIT(SPDIF_BIPHASE_ENABLE)
+#define SPDIF_BIPHASE_IDLE_MASK		BIT(SPDIF_BIPHASE_IDLE)
+
+enum {
+	STI_SAS_DAI_SPDIF_OUT,
+	STI_SAS_DAI_ANALOG_OUT,
+};
+
+static const struct reg_default stih416_sas_reg_defaults[] = {
+	{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
+	{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
+};
+
+static const struct reg_default stih407_sas_reg_defaults[] = {
+	{ STIH416_AUDIO_DAC_CTRL, 0x000000000 },
+	{ STIH416_AUDIO_GLUE_CTRL, 0x00000040 },
+};
+
+struct sti_dac_audio {
+	struct regmap *regmap;
+	struct regmap *virt_regmap;
+	struct regmap_field  **field;
+	struct reset_control *rst;
+	int mclk;
+};
+
+struct sti_spdif_audio {
+	struct regmap *regmap;
+	struct regmap_field  **field;
+	int mclk;
+};
+
+/* device data structure */
+struct sti_sas_dev_data {
+	const int chipid; /* IC version */
+	const struct regmap_config *regmap;
+	const struct snd_soc_dai_ops *dac_ops;  /* DAC function callbacks */
+	const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */
+	const int num_dapm_widgets; /* dapms declaration */
+	const struct snd_soc_dapm_route *dapm_routes; /* route declaration */
+	const int num_dapm_routes; /* route declaration */
+};
+
+/* driver data structure */
+struct sti_sas_data {
+	struct device *dev;
+	const struct sti_sas_dev_data *dev_data;
+	struct sti_dac_audio dac;
+	struct sti_spdif_audio spdif;
+};
+
+/* Read a register from the sysconf reg bank */
+static int sti_sas_read_reg(void *context, unsigned int reg,
+			    unsigned int *value)
+{
+	struct sti_sas_data *drvdata = context;
+	int status;
+	u32 val;
+
+	status = regmap_read(drvdata->dac.regmap, reg, &val);
+	*value = (unsigned int)val;
+
+	return status;
+}
+
+/* Read a register from the sysconf reg bank */
+static int sti_sas_write_reg(void *context, unsigned int reg,
+			     unsigned int value)
+{
+	struct sti_sas_data *drvdata = context;
+	int status;
+
+	status = regmap_write(drvdata->dac.regmap, reg, value);
+
+	return status;
+}
+
+static int  sti_sas_init_sas_registers(struct snd_soc_codec *codec,
+				       struct sti_sas_data *data)
+{
+	int ret;
+	/*
+	 * DAC and SPDIF are activated by default
+	 * put them in IDLE to save power
+	 */
+
+	/* Initialise bi-phase formatter to disabled */
+	ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
+				  SPDIF_BIPHASE_ENABLE_MASK, 0);
+
+	if (!ret)
+		/* Initialise bi-phase formatter idle value to 0 */
+		ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
+					  SPDIF_BIPHASE_IDLE_MASK, 0);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to update SPDIF registers");
+		return ret;
+	}
+
+	/* Init DAC configuration */
+	switch (data->dev_data->chipid) {
+	case CHIPID_STIH407:
+		/* init configuration */
+		ret =  snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+					   STIH407_DAC_STANDBY_MASK,
+					   STIH407_DAC_STANDBY_MASK);
+
+		if (!ret)
+			ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+						  STIH407_DAC_STANDBY_ANA_MASK,
+						  STIH407_DAC_STANDBY_ANA_MASK);
+		if (!ret)
+			ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+						  STIH407_DAC_SOFTMUTE_MASK,
+						  STIH407_DAC_SOFTMUTE_MASK);
+		break;
+	case CHIPID_STIH416:
+		ret =  snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
+					   STIH416_DAC_NOT_STANDBY_MASK, 0);
+		if (!ret)
+			ret =  snd_soc_update_bits(codec,
+						   STIH416_AUDIO_DAC_CTRL,
+						   STIH416_DAC_ANA_NOT_PWR, 0);
+		if (!ret)
+			ret =  snd_soc_update_bits(codec,
+						   STIH416_AUDIO_DAC_CTRL,
+						   STIH416_DAC_NOT_PNDBG_MASK,
+						   0);
+		if (!ret)
+			ret =  snd_soc_update_bits(codec,
+						   STIH416_AUDIO_DAC_CTRL,
+						   STIH416_DAC_SOFTMUTE_MASK,
+						   STIH416_DAC_SOFTMUTE_MASK);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to update DAC registers");
+		return ret;
+	}
+
+	return ret;
+}
+
+/*
+ * DAC
+ */
+static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	/* Sanity check only */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupporter master mask 0x%x\n",
+			__func__, fmt & SND_SOC_DAIFMT_MASTER_MASK);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int stih416_dac_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
+	struct sti_dac_audio *dac = &drvdata->dac;
+
+	/* Get reset control */
+	dac->rst = devm_reset_control_get(codec->dev, "dac_rst");
+	if (IS_ERR(dac->rst)) {
+		dev_err(dai->codec->dev,
+			"%s: ERROR: DAC reset control not defined !\n",
+			__func__);
+		dac->rst = NULL;
+		return -EFAULT;
+	}
+	/* Put the DAC into reset */
+	reset_control_assert(dac->rst);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget stih416_sas_dapm_widgets[] = {
+	SND_SOC_DAPM_PGA("DAC bandgap", STIH416_AUDIO_DAC_CTRL,
+			 STIH416_DAC_NOT_PNDBG_MASK, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH416_AUDIO_DAC_CTRL,
+			     STIH416_DAC_ANA_NOT_PWR, 0, NULL, 0),
+	SND_SOC_DAPM_DAC("DAC standby",  "dac_p", STIH416_AUDIO_DAC_CTRL,
+			 STIH416_DAC_NOT_STANDBY, 0),
+	SND_SOC_DAPM_OUTPUT("DAC Output"),
+};
+
+static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH407_AUDIO_DAC_CTRL,
+			     STIH407_DAC_STANDBY_ANA, 1, NULL, 0),
+	SND_SOC_DAPM_DAC("DAC standby",  "dac_p", STIH407_AUDIO_DAC_CTRL,
+			 STIH407_DAC_STANDBY, 1),
+	SND_SOC_DAPM_OUTPUT("DAC Output"),
+};
+
+static const struct snd_soc_dapm_route stih416_sas_route[] = {
+	{"DAC Output", NULL, "DAC bandgap"},
+	{"DAC Output", NULL, "DAC standby ana"},
+	{"DAC standby ana", NULL, "DAC standby"},
+};
+
+static const struct snd_soc_dapm_route stih407_sas_route[] = {
+	{"DAC Output", NULL, "DAC standby ana"},
+	{"DAC standby ana", NULL, "DAC standby"},
+};
+
+static int stih416_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	if (mute) {
+		return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
+					    STIH416_DAC_SOFTMUTE_MASK,
+					    STIH416_DAC_SOFTMUTE_MASK);
+	} else {
+		return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
+					    STIH416_DAC_SOFTMUTE_MASK, 0);
+	}
+}
+
+static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	if (mute) {
+		return snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+					    STIH407_DAC_SOFTMUTE_MASK,
+					    STIH407_DAC_SOFTMUTE_MASK);
+	} else {
+		return snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+					    STIH407_DAC_SOFTMUTE_MASK,
+					    0);
+	}
+}
+
+/*
+ * SPDIF
+ */
+static int sti_sas_spdif_set_fmt(struct snd_soc_dai *dai,
+				 unsigned int fmt)
+{
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupporter master mask 0x%x\n",
+			__func__, fmt & SND_SOC_DAIFMT_MASTER_MASK);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * sti_sas_spdif_trigger:
+ * Trigger function is used to ensure that BiPhase Formater is disabled
+ * before CPU dai is stopped.
+ * This is mandatory to avoid that BPF is stalled
+ */
+static int sti_sas_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+				 struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		return snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
+					    SPDIF_BIPHASE_ENABLE_MASK,
+					    SPDIF_BIPHASE_ENABLE_MASK);
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		return snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
+					    SPDIF_BIPHASE_ENABLE_MASK,
+					    0);
+	default:
+		return -EINVAL;
+	}
+}
+
+static bool sti_sas_volatile_register(struct device *dev, unsigned int reg)
+{
+	if (reg == STIH407_AUDIO_GLUE_CTRL)
+		return true;
+
+	return false;
+}
+
+/*
+ * CODEC DAIS
+ */
+
+/*
+ * sti_sas_set_sysclk:
+ * get MCLK input frequency to check that MCLK-FS ratio is coherent
+ */
+static int sti_sas_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			      unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
+
+	if (dir == SND_SOC_CLOCK_OUT)
+		return 0;
+
+	if (clk_id != 0)
+		return -EINVAL;
+
+	switch (dai->id) {
+	case STI_SAS_DAI_SPDIF_OUT:
+		drvdata->spdif.mclk = freq;
+		break;
+
+	case STI_SAS_DAI_ANALOG_OUT:
+		drvdata->dac.mclk = freq;
+		break;
+	}
+
+	return 0;
+}
+
+static int sti_sas_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	switch (dai->id) {
+	case STI_SAS_DAI_SPDIF_OUT:
+		if ((drvdata->spdif.mclk / runtime->rate) != 128) {
+			dev_err(codec->dev, "unexpected mclk-fs ratio");
+			return -EINVAL;
+		}
+		break;
+	case STI_SAS_DAI_ANALOG_OUT:
+		if ((drvdata->dac.mclk / runtime->rate) != 256) {
+			dev_err(codec->dev, "unexpected mclk-fs ratio");
+			return -EINVAL;
+		}
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops stih416_dac_ops = {
+	.set_fmt = sti_sas_dac_set_fmt,
+	.mute_stream = stih416_sas_dac_mute,
+	.prepare = sti_sas_prepare,
+	.set_sysclk = sti_sas_set_sysclk,
+};
+
+static const struct snd_soc_dai_ops stih407_dac_ops = {
+	.set_fmt = sti_sas_dac_set_fmt,
+	.mute_stream = stih407_sas_dac_mute,
+	.prepare = sti_sas_prepare,
+	.set_sysclk = sti_sas_set_sysclk,
+};
+
+static const struct regmap_config stih407_sas_regmap = {
+	.reg_bits = 32,
+	.val_bits = 32,
+
+	.max_register = STIH407_AUDIO_DAC_CTRL,
+	.reg_defaults = stih407_sas_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(stih407_sas_reg_defaults),
+	.volatile_reg = sti_sas_volatile_register,
+	.cache_type = REGCACHE_RBTREE,
+	.reg_read = sti_sas_read_reg,
+	.reg_write = sti_sas_write_reg,
+};
+
+static const struct regmap_config stih416_sas_regmap = {
+	.reg_bits = 32,
+	.val_bits = 32,
+
+	.max_register = STIH416_AUDIO_DAC_CTRL,
+	.reg_defaults = stih416_sas_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(stih416_sas_reg_defaults),
+	.volatile_reg = sti_sas_volatile_register,
+	.cache_type = REGCACHE_RBTREE,
+	.reg_read = sti_sas_read_reg,
+	.reg_write = sti_sas_write_reg,
+};
+
+static const struct sti_sas_dev_data stih416_data = {
+	.chipid = CHIPID_STIH416,
+	.regmap = &stih416_sas_regmap,
+	.dac_ops = &stih416_dac_ops,
+	.dapm_widgets = stih416_sas_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(stih416_sas_dapm_widgets),
+	.dapm_routes =	stih416_sas_route,
+	.num_dapm_routes = ARRAY_SIZE(stih416_sas_route),
+};
+
+static const struct sti_sas_dev_data stih407_data = {
+	.chipid = CHIPID_STIH407,
+	.regmap = &stih407_sas_regmap,
+	.dac_ops = &stih407_dac_ops,
+	.dapm_widgets = stih407_sas_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(stih407_sas_dapm_widgets),
+	.dapm_routes =	stih407_sas_route,
+	.num_dapm_routes = ARRAY_SIZE(stih407_sas_route),
+};
+
+static struct snd_soc_dai_driver sti_sas_dai[] = {
+	{
+		.name = "sas-dai-spdif-out",
+		.id = STI_SAS_DAI_SPDIF_OUT,
+		.playback = {
+			.stream_name = "spdif_p",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+				 SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 |
+				 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S32_LE,
+		},
+		.ops = (struct snd_soc_dai_ops[]) {
+			{
+				.set_fmt = sti_sas_spdif_set_fmt,
+				.trigger = sti_sas_spdif_trigger,
+				.set_sysclk = sti_sas_set_sysclk,
+				.prepare = sti_sas_prepare,
+			}
+		},
+	},
+	{
+		.name = "sas-dai-dac",
+		.id = STI_SAS_DAI_ANALOG_OUT,
+		.playback = {
+			.stream_name = "dac_p",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S32_LE,
+		},
+	},
+};
+
+#ifdef CONFIG_PM_SLEEP
+static int sti_sas_resume(struct snd_soc_codec *codec)
+{
+	struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
+
+	return sti_sas_init_sas_registers(codec, drvdata);
+}
+#else
+#define sti_sas_resume NULL
+#endif
+
+static int sti_sas_codec_probe(struct snd_soc_codec *codec)
+{
+	struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
+	int ret;
+
+	ret = sti_sas_init_sas_registers(codec, drvdata);
+
+	return ret;
+}
+
+static struct snd_soc_codec_driver sti_sas_driver = {
+	.probe = sti_sas_codec_probe,
+	.resume = sti_sas_resume,
+};
+
+static const struct of_device_id sti_sas_dev_match[] = {
+	{
+		.compatible = "st,stih416-sas-codec",
+		.data = &stih416_data,
+	},
+	{
+		.compatible = "st,stih407-sas-codec",
+		.data = &stih407_data,
+	},
+	{},
+};
+
+static int sti_sas_driver_probe(struct platform_device *pdev)
+{
+	struct device_node *pnode = pdev->dev.of_node;
+	struct sti_sas_data *drvdata;
+	const struct of_device_id *of_id;
+
+	/* Allocate device structure */
+	drvdata = devm_kzalloc(&pdev->dev, sizeof(struct sti_sas_data),
+			       GFP_KERNEL);
+	if (!drvdata)
+		return -ENOMEM;
+
+	/* Populate data structure depending on compatibility */
+	of_id = of_match_node(sti_sas_dev_match, pnode);
+	if (!of_id->data) {
+		dev_err(&pdev->dev, "data associated to device is missing");
+		return -EINVAL;
+	}
+
+	drvdata->dev_data = (struct sti_sas_dev_data *)of_id->data;
+
+	/* Initialise device structure */
+	drvdata->dev = &pdev->dev;
+
+	/* Request the DAC & SPDIF registers memory region */
+	drvdata->dac.virt_regmap = devm_regmap_init(&pdev->dev, NULL, drvdata,
+						    drvdata->dev_data->regmap);
+	if (IS_ERR(drvdata->dac.virt_regmap)) {
+		dev_err(&pdev->dev, "audio registers not enabled\n");
+		return PTR_ERR(drvdata->dac.virt_regmap);
+	}
+
+	/* Request the syscon region */
+	drvdata->dac.regmap =
+		syscon_regmap_lookup_by_phandle(pnode, "st,syscfg");
+	if (IS_ERR(drvdata->dac.regmap)) {
+		dev_err(&pdev->dev, "syscon registers not available\n");
+		return PTR_ERR(drvdata->dac.regmap);
+	}
+	drvdata->spdif.regmap = drvdata->dac.regmap;
+
+	/* Set DAC dai probe */
+	if (drvdata->dev_data->chipid == CHIPID_STIH416)
+		sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].probe = stih416_dac_probe;
+
+	sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops;
+
+	/* Set dapms*/
+	sti_sas_driver.dapm_widgets = drvdata->dev_data->dapm_widgets;
+	sti_sas_driver.num_dapm_widgets = drvdata->dev_data->num_dapm_widgets;
+
+	sti_sas_driver.dapm_routes = drvdata->dev_data->dapm_routes;
+	sti_sas_driver.num_dapm_routes = drvdata->dev_data->num_dapm_routes;
+
+	/* Store context */
+	dev_set_drvdata(&pdev->dev, drvdata);
+
+	return snd_soc_register_codec(&pdev->dev, &sti_sas_driver,
+					sti_sas_dai,
+					ARRAY_SIZE(sti_sas_dai));
+}
+
+static int sti_sas_driver_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver sti_sas_platform_driver = {
+	.driver = {
+		.name = "sti-sas-codec",
+		.of_match_table = sti_sas_dev_match,
+	},
+	.probe = sti_sas_driver_probe,
+	.remove = sti_sas_driver_remove,
+};
+
+module_platform_driver(sti_sas_platform_driver);
+
+MODULE_DESCRIPTION("audio codec for STMicroelectronics sti platforms");
+MODULE_AUTHOR("Arnaud.pouliquen@st.com");
+MODULE_LICENSE("GPL v2");