Merge tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This time it's a relatively calm update batch, but the amount isn't
  too small in the end.  Here we go over some highlights:

  ALSA core:
   - One major change is the support of nonatomic PCM operations.  This
     allows the trigger and other callbacks to call schedule(), which
     would be useful for mailbox type communications.  Already some
     drivers (Digigram ones) have been converted to use together with
     threaded irqs as an example.
   - Improvement / fixes of DSD PCM format support

  HD-audio:
   - Large volume of rewrites are found in Realtek codec driver for
     converting Dell and HP quirks to generic forms.
   - Inverted dmic code cleanup from David.
   - Realtek COEF access has been optimized.
   - Now HD-audio jack infrastructure allows multiple callbacks, which
     fixes / simplifies the jack-dependent power controls on STAC/IDT
     and VIA codecs.
   - Many additional device-specific fixups as usual
   - A few deadcode cleanups, CA0132 code cleanup, etc.

  ASoC:
   - More componentization work from Lars-Peter, this time mainly
     cleaning up the suspend and bias level transition callbacks.
   - Real system support for the Intel drivers and a bunch of fixes and
     enhancements for the associated CODEC drivers, this is going to
     need a lot quirks over time due to the lack of any firmware
     description of the boards.
   - Jack detect support for simple card from Dylan Reid.
   - A bunch of small fixes and enhancements for the Freescale drivers.
   - New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32,
     Everest Semiconductor ES8328 and Freescale cards using the ASRC in
     newer i.MX processors.
   - A few simple-card fixes, mostly cleanups but also a fix for
     interaction between GPIO 0 and simple-card.

  Misc:
   - Virtuoso / Oxygen updates by Clemens
   - USB-audio: Yamaha MOTIF XF MIDI port name fixes
   - Conversion of kernel messages to standard dev_*() in ctxfi driver"

* tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (251 commits)
  ASoC: mc13783: Ensure we only try to dereference valid of_nodes
  ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_txctrl
  ALSA: hda - Add dock port support to Thinkpad L440 (71aa:501e)
  ALSA: Allow pass NULL dev for snd_pci_quirk_lookup()
  ASoC: imx-es8328: Fix of_node_put() call with uninitialized object
  ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb()
  ASoC: simple-card: Initialize headphone and mic GPIO numbers
  ASoC: imx-es8328: Fix missing return code in imx_es8328_probe()
  ALSA: hda - Add dock support for Thinkpad T440 (17aa:2212)
  ALSA: usb: caiaq: check for cdev->n_streams > 1
  ASoC: 88pm860x-codec: Fix possibly missing string termination
  ASoC: core: fix use after free in snd_soc_remove_platform()
  ASoC: soc-dapm: fix use after free
  ALSA: hda - Make the inv dmic handling for Realtek use generic parser
  ALSA: hda - Add Inverted Internal mic for Samsung Ativ book 9 (NP900X3G)
  ALSA: hda - Add inverted internal mic for Asus Aspire 4830T
  ASoC: Intel: byt-rt5640: fix coccinelle warnings
  ASoC: fsl_esai doc: Add "fsl,vf610-esai" as compatible string
  ASoC: da732x: Remove unnecessary KERN_ERR in pr_err()
  ASoC: simple-card: Fix detect gpio documentation.
  ...

11 files changed, 977 insertions, 54 deletions

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f3012b645b51..081e406b3713 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -240,6 +240,18 @@ config SND_SOC_IMX_WM8962
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a wm8962 codec.
 
+config SND_SOC_IMX_ES8328
+	tristate "SoC Audio support for i.MX boards with the ES8328 codec"
+	depends on OF && (I2C || SPI)
+	select SND_SOC_ES8328_I2C if I2C
+	select SND_SOC_ES8328_SPI if SPI_MASTER
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_FSL_SSI
+	help
+	  Say Y if you want to add support for the ES8328 audio codec connected
+	  via SSI/I2S over either SPI or I2C.
+
 config SND_SOC_IMX_SGTL5000
 	tristate "SoC Audio support for i.MX boards with sgtl5000"
 	depends on OF && I2C
@@ -268,6 +280,20 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
+config SND_SOC_FSL_ASOC_CARD
+	tristate "Generic ASoC Sound Card with ASRC support"
+	depends on OF && I2C
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_ESAI
+	select SND_SOC_FSL_SAI
+	select SND_SOC_FSL_SSI
+	help
+	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
+	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
+	 and SGTL5000.
+	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
+
 endif # SND_IMX_SOC
 
 endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 9ff59267eac9..d28dc25c9375 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
 # Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o
 snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
 snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-y := fsl_ssi.o
@@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-esai-objs := fsl_esai.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
 obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
 obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
@@ -50,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
 snd-soc-phycore-ac97-objs := phycore-ac97.o
 snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-es8328-objs := imx-es8328.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
 snd-soc-imx-spdif-objs := imx-spdif.o
@@ -59,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
 obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
+obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
 obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
new file mode 100644
index 000000000000..007c772f3cef
--- /dev/null
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -0,0 +1,574 @@
+/*
+ * Freescale Generic ASoC Sound Card driver with ASRC
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "fsl_esai.h"
+#include "fsl_sai.h"
+#include "imx-audmux.h"
+
+#include "../codecs/sgtl5000.h"
+#include "../codecs/wm8962.h"
+
+#define RX 0
+#define TX 1
+
+/* Default DAI format without Master and Slave flag */
+#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
+
+/**
+ * CODEC private data
+ *
+ * @mclk_freq: Clock rate of MCLK
+ * @mclk_id: MCLK (or main clock) id for set_sysclk()
+ * @fll_id: FLL (or secordary clock) id for set_sysclk()
+ * @pll_id: PLL id for set_pll()
+ */
+struct codec_priv {
+	unsigned long mclk_freq;
+	u32 mclk_id;
+	u32 fll_id;
+	u32 pll_id;
+};
+
+/**
+ * CPU private data
+ *
+ * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
+ * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
+ * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ *
+ * Note: [1] for tx and [0] for rx
+ */
+struct cpu_priv {
+	unsigned long sysclk_freq[2];
+	u32 sysclk_dir[2];
+	u32 sysclk_id[2];
+};
+
+/**
+ * Freescale Generic ASOC card private data
+ *
+ * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * @pdev: platform device pointer
+ * @codec_priv: CODEC private data
+ * @cpu_priv: CPU private data
+ * @card: ASoC card structure
+ * @sample_rate: Current sample rate
+ * @sample_format: Current sample format
+ * @asrc_rate: ASRC sample rate used by Back-Ends
+ * @asrc_format: ASRC sample format used by Back-Ends
+ * @dai_fmt: DAI format between CPU and CODEC
+ * @name: Card name
+ */
+
+struct fsl_asoc_card_priv {
+	struct snd_soc_dai_link dai_link[3];
+	struct platform_device *pdev;
+	struct codec_priv codec_priv;
+	struct cpu_priv cpu_priv;
+	struct snd_soc_card card;
+	u32 sample_rate;
+	u32 sample_format;
+	u32 asrc_rate;
+	u32 asrc_format;
+	u32 dai_fmt;
+	char name[32];
+};
+
+/**
+ * This dapm route map exsits for DPCM link only.
+ * The other routes shall go through Device Tree.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"CPU-Playback",  NULL, "ASRC-Playback"},
+	{"Playback",  NULL, "CPU-Playback"},
+	{"ASRC-Capture",  NULL, "CPU-Capture"},
+	{"CPU-Capture",  NULL, "Capture"},
+};
+
+/* Add all possible widgets into here without being redundant */
+static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("AMIC", NULL),
+	SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	struct cpu_priv *cpu_priv = &priv->cpu_priv;
+	struct device *dev = rtd->card->dev;
+	int ret;
+
+	priv->sample_rate = params_rate(params);
+	priv->sample_format = params_format(params);
+
+	if (priv->card.set_bias_level)
+		return 0;
+
+	/* Specific configurations of DAIs starts from here */
+	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+				     cpu_priv->sysclk_freq[tx],
+				     cpu_priv->sysclk_dir[tx]);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk for cpu dai\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops fsl_asoc_card_ops = {
+	.hw_params = fsl_asoc_card_hw_params,
+};
+
+static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+			      struct snd_pcm_hw_params *params)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_interval *rate;
+	struct snd_mask *mask;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	rate->max = rate->min = priv->asrc_rate;
+
+	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	snd_mask_none(mask);
+	snd_mask_set(mask, priv->asrc_format);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
+	/* Default ASoC DAI Link*/
+	{
+		.name = "HiFi",
+		.stream_name = "HiFi",
+		.ops = &fsl_asoc_card_ops,
+	},
+	/* DPCM Link between Front-End and Back-End (Optional) */
+	{
+		.name = "HiFi-ASRC-FE",
+		.stream_name = "HiFi-ASRC-FE",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.dynamic = 1,
+	},
+	{
+		.name = "HiFi-ASRC-BE",
+		.stream_name = "HiFi-ASRC-BE",
+		.platform_name = "snd-soc-dummy",
+		.be_hw_params_fixup = be_hw_params_fixup,
+		.ops = &fsl_asoc_card_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
+					enum snd_soc_bias_level level)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	unsigned int pll_out;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
+			break;
+
+		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+			pll_out = priv->sample_rate * 384;
+		else
+			pll_out = priv->sample_rate * 256;
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
+					  codec_priv->mclk_id,
+					  codec_priv->mclk_freq, pll_out);
+		if (ret) {
+			dev_err(dev, "failed to start FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
+					     pll_out, SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
+			break;
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+					     codec_priv->mclk_freq,
+					     SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
+		if (ret) {
+			dev_err(dev, "failed to stop FLL: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_audmux_init(struct device_node *np,
+				     struct fsl_asoc_card_priv *priv)
+{
+	struct device *dev = &priv->pdev->dev;
+	u32 int_ptcr = 0, ext_ptcr = 0;
+	int int_port, ext_port;
+	int ret;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		return ret;
+	}
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		return ret;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the AUDMUX API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+
+	/*
+	 * Use asynchronous mode (6 wires) for all cases.
+	 * If only 4 wires are needed, just set SSI into
+	 * synchronous mode and enable 4 PADs in IOMUX.
+	 */
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Asynchronous mode can not be set along with RCLKDIR */
+	ret = imx_audmux_v2_configure_port(int_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk in %s\n", __func__);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_probe(struct platform_device *pdev)
+{
+	struct device_node *cpu_np, *codec_np, *asrc_np;
+	struct device_node *np = pdev->dev.of_node;
+	struct platform_device *asrc_pdev = NULL;
+	struct platform_device *cpu_pdev;
+	struct fsl_asoc_card_priv *priv;
+	struct i2c_client *codec_dev;
+	struct clk *codec_clk;
+	u32 width;
+	int ret;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
+	/* Give a chance to old DT binding */
+	if (!cpu_np)
+		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!cpu_np || !codec_np) {
+		dev_err(&pdev->dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	cpu_pdev = of_find_device_by_node(cpu_np);
+	if (!cpu_pdev) {
+		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	codec_dev = of_find_i2c_device_by_node(codec_np);
+	if (!codec_dev) {
+		dev_err(&pdev->dev, "failed to find codec platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
+	if (asrc_np)
+		asrc_pdev = of_find_device_by_node(asrc_np);
+
+	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
+	codec_clk = clk_get(&codec_dev->dev, NULL);
+	if (!IS_ERR(codec_clk)) {
+		priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+		clk_put(codec_clk);
+	}
+
+	/* Default sample rate and format, will be updated in hw_params() */
+	priv->sample_rate = 44100;
+	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
+
+	/* Assign a default DAI format, and allow each card to overwrite it */
+	priv->dai_fmt = DAI_FMT_BASE;
+
+	/* Diversify the card configurations */
+	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
+		priv->card.set_bias_level = NULL;
+		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
+		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
+		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
+		priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
+		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
+		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
+		priv->codec_priv.pll_id = WM8962_FLL;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else {
+		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
+		return -EINVAL;
+	}
+
+	/* Common settings for corresponding Freescale CPU DAI driver */
+	if (strstr(cpu_np->name, "ssi")) {
+		/* Only SSI needs to configure AUDMUX */
+		ret = fsl_asoc_card_audmux_init(np, priv);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to init audmux\n");
+			goto asrc_fail;
+		}
+	} else if (strstr(cpu_np->name, "esai")) {
+		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
+		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
+	} else if (strstr(cpu_np->name, "sai")) {
+		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
+		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
+	}
+
+	sprintf(priv->name, "%s-audio", codec_dev->name);
+
+	/* Initialize sound card */
+	priv->pdev = pdev;
+	priv->card.dev = &pdev->dev;
+	priv->card.name = priv->name;
+	priv->card.dai_link = priv->dai_link;
+	priv->card.dapm_routes = audio_map;
+	priv->card.late_probe = fsl_asoc_card_late_probe;
+	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
+	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
+	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
+
+	memcpy(priv->dai_link, fsl_asoc_card_dai,
+	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+	/* Normal DAI Link */
+	priv->dai_link[0].cpu_of_node = cpu_np;
+	priv->dai_link[0].codec_of_node = codec_np;
+	priv->dai_link[0].codec_dai_name = codec_dev->name;
+	priv->dai_link[0].platform_of_node = cpu_np;
+	priv->dai_link[0].dai_fmt = priv->dai_fmt;
+	priv->card.num_links = 1;
+
+	if (asrc_pdev) {
+		/* DPCM DAI Links only if ASRC exsits */
+		priv->dai_link[1].cpu_of_node = asrc_np;
+		priv->dai_link[1].platform_of_node = asrc_np;
+		priv->dai_link[2].codec_dai_name = codec_dev->name;
+		priv->dai_link[2].codec_of_node = codec_np;
+		priv->dai_link[2].cpu_of_node = cpu_np;
+		priv->dai_link[2].dai_fmt = priv->dai_fmt;
+		priv->card.num_links = 3;
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
+					   &priv->asrc_rate);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto asrc_fail;
+		}
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto asrc_fail;
+		}
+
+		if (width == 24)
+			priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
+		else
+			priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
+	}
+
+	/* Finish card registering */
+	platform_set_drvdata(pdev, priv);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+asrc_fail:
+	of_node_put(asrc_np);
+fail:
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	return ret;
+}
+
+static const struct of_device_id fsl_asoc_card_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-cs42888", },
+	{ .compatible = "fsl,imx-audio-sgtl5000", },
+	{ .compatible = "fsl,imx-audio-wm8962", },
+	{}
+};
+
+static struct platform_driver fsl_asoc_card_driver = {
+	.probe = fsl_asoc_card_probe,
+	.driver = {
+		.name = "fsl-asoc-card",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = fsl_asoc_card_dt_ids,
+	},
+};
+module_platform_driver(fsl_asoc_card_driver);
+
+MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
+MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
+MODULE_ALIAS("platform:fsl-asoc-card");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 822110420b71..3b145313f93e 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_asrc_regmap_config = {
+static const struct regmap_config fsl_asrc_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
 
 	asrc_priv->paddr = res->start;
 
-	/* Register regmap and let it prepare core clock */
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs,
 						      &fsl_asrc_regmap_config);
 	if (IS_ERR(asrc_priv->regmap)) {
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a3b29ed84963..8bcdfda09d7a 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -37,6 +37,7 @@
  * @fsysclk: system clock source to derive HCK, SCK and FS
  * @fifo_depth: depth of tx/rx FIFO
  * @slot_width: width of each DAI slot
+ * @slots: number of slots
  * @hck_rate: clock rate of desired HCKx clock
  * @sck_rate: clock rate of desired SCKx clock
  * @hck_dir: the direction of HCKx pads
@@ -55,6 +56,7 @@ struct fsl_esai {
 	struct clk *fsysclk;
 	u32 fifo_depth;
 	u32 slot_width;
+	u32 slots;
 	u32 hck_rate[2];
 	u32 sck_rate[2];
 	bool hck_dir[2];
@@ -362,6 +364,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
 
 	esai_priv->slot_width = slot_width;
+	esai_priv->slots = slots;
 
 	return 0;
 }
@@ -509,10 +512,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 width = snd_pcm_format_width(params_format(params));
 	u32 channels = params_channels(params);
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 	u32 bclk, mask, val;
 	int ret;
 
-	bclk = params_rate(params) * esai_priv->slot_width * 2;
+	bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots;
 
 	ret = fsl_esai_set_bclk(dai, tx, bclk);
 	if (ret)
@@ -529,7 +533,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
 	      (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
 	val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
-	     (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels));
+	     (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins));
 
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
 
@@ -564,6 +568,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u8 i, channels = substream->runtime->channels;
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -578,7 +583,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
-				   tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels));
+				   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -705,7 +710,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_esai_regmap_config = {
+static const struct regmap_config fsl_esai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -731,9 +736,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	esai_priv->pdev = pdev;
 	strcpy(esai_priv->name, np->name);
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
@@ -781,6 +783,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	/* Set a default slot size */
 	esai_priv->slot_width = 32;
 
+	/* Set a default slot number */
+	esai_priv->slots = 2;
+
 	/* Set a default master/slave state */
 	esai_priv->slave_mode = true;
 
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 75e14033e8d8..91a550f4a10d 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -130,8 +130,8 @@
 #define ESAI_xFCR_RE_WIDTH	4
 #define ESAI_xFCR_TE_MASK	(((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
 #define ESAI_xFCR_RE_MASK	(((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
-#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK)
-#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK)
+#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK)
+#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK)
 #define ESAI_xFCR_xFR_SHIFT	1
 #define ESAI_xFCR_xFR_MASK	(1 << ESAI_xFCR_xFR_SHIFT)
 #define ESAI_xFCR_xFR		(1 << ESAI_xFCR_xFR_SHIFT)
@@ -272,8 +272,8 @@
 #define ESAI_xCR_RE_WIDTH	4
 #define ESAI_xCR_TE_MASK	(((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
 #define ESAI_xCR_RE_MASK	(((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
-#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK)
-#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK)
+#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK)
+#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK)
 
 /*
  * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index faa049797897..7eeb1dd8ce27 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	bool tx = fsl_dir == FSL_FMT_TRANSMITTER;
 	u32 val_cr2 = 0, val_cr4 = 0;
 
-	if (!sai->big_endian_data)
+	if (!sai->is_lsb_first)
 		val_cr4 |= FSL_SAI_CR4_MF;
 
 	/* DAI mode */
@@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
 	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
 
-	if (sai->big_endian_data)
+	if (sai->is_lsb_first)
 		val_cr5 |= FSL_SAI_CR5_FBT(0);
 	else
 		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
@@ -330,13 +330,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 	u32 xcsr, count = 100;
 
 	/*
-	 * The transmitter bit clock and frame sync are to be
-	 * used by both the transmitter and receiver.
+	 * Asynchronous mode: Clear SYNC for both Tx and Rx.
+	 * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
+	 * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
 	 */
-	regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
-			   ~FSL_SAI_CR2_SYNC);
+	regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
-			   FSL_SAI_CR2_SYNC);
+			   sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
 
 	/*
 	 * It is recommended that the transmitter is the last enabled
@@ -437,8 +437,13 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
 
-	regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0);
-	regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0);
+	/* Software Reset for both Tx and Rx */
+	regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
+	regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+	/* Clear SR bit to finish the reset */
+	regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
+	regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+
 	regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
 			   FSL_SAI_MAXBURST_TX * 2);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
@@ -539,7 +544,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_sai_regmap_config = {
+static const struct regmap_config fsl_sai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -568,11 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
 		sai->sai_on_imx = true;
 
-	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
-	if (sai->big_endian_regs)
-		fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
-	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+	sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	base = devm_ioremap_resource(&pdev->dev, res);
@@ -621,6 +622,33 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	/* Sync Tx with Rx as default by following old DT binding */
+	sai->synchronous[RX] = true;
+	sai->synchronous[TX] = false;
+	fsl_sai_dai.symmetric_rates = 1;
+	fsl_sai_dai.symmetric_channels = 1;
+	fsl_sai_dai.symmetric_samplebits = 1;
+
+	if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) &&
+	    of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+		/* error out if both synchronous and asynchronous are present */
+		dev_err(&pdev->dev, "invalid binding for synchronous mode\n");
+		return -EINVAL;
+	}
+
+	if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) {
+		/* Sync Rx with Tx */
+		sai->synchronous[RX] = false;
+		sai->synchronous[TX] = true;
+	} else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+		/* Discard all settings for asynchronous mode */
+		sai->synchronous[RX] = false;
+		sai->synchronous[TX] = false;
+		fsl_sai_dai.symmetric_rates = 0;
+		fsl_sai_dai.symmetric_channels = 0;
+		fsl_sai_dai.symmetric_samplebits = 0;
+	}
+
 	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
 	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
 	sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 0e6c9f595d75..34667209b607 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -48,6 +48,7 @@
 /* SAI Transmit/Recieve Control Register */
 #define FSL_SAI_CSR_TERE	BIT(31)
 #define FSL_SAI_CSR_FR		BIT(25)
+#define FSL_SAI_CSR_SR		BIT(24)
 #define FSL_SAI_CSR_xF_SHIFT	16
 #define FSL_SAI_CSR_xF_W_SHIFT	18
 #define FSL_SAI_CSR_xF_MASK	(0x1f << FSL_SAI_CSR_xF_SHIFT)
@@ -131,13 +132,16 @@ struct fsl_sai {
 	struct clk *bus_clk;
 	struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
 
-	bool big_endian_regs;
-	bool big_endian_data;
+	bool is_lsb_first;
 	bool is_dsp_mode;
 	bool sai_on_imx;
+	bool synchronous[2];
 
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 };
 
+#define TX 1
+#define RX 0
+
 #endif /* __FSL_SAI_H */
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 70acfe4a9bd5..9b791621294c 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -15,7 +15,6 @@
 
 #include <linux/bitrev.h>
 #include <linux/clk.h>
-#include <linux/clk-private.h>
 #include <linux/module.h>
 #include <linux/of_address.h>
 #include <linux/of_device.h>
@@ -1040,7 +1039,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_spdif_regmap_config = {
+static const struct regmap_config fsl_spdif_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -1184,9 +1183,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
 	spdif_priv->cpu_dai_drv.name = spdif_priv->name;
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index de6ab06f58a5..e6955170dc42 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -169,6 +169,7 @@ struct fsl_ssi_private {
 	u8 i2s_mode;
 	bool use_dma;
 	bool use_dual_fifo;
+	bool has_ipg_clk_name;
 	unsigned int fifo_depth;
 	struct fsl_ssi_rxtx_reg_val rxtx_reg_val;
 
@@ -259,6 +260,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
 		SND_SOC_DAIFMT_CBS_CFS;
 }
 
+static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private)
+{
+	return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+		SND_SOC_DAIFMT_CBM_CFS;
+}
 /**
  * fsl_ssi_isr: SSI interrupt handler
  *
@@ -525,6 +531,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private =
 		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	int ret;
+
+	ret = clk_prepare_enable(ssi_private->clk);
+	if (ret)
+		return ret;
 
 	/* When using dual fifo mode, it is safer to ensure an even period
 	 * size. If appearing to an odd number while DMA always starts its
@@ -539,6 +550,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 }
 
 /**
+ * fsl_ssi_shutdown: shutdown the SSI
+ *
+ */
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_ssi_private *ssi_private =
+		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+
+	clk_disable_unprepare(ssi_private->clk);
+
+}
+
+/**
  * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock
  *
  * Note: This function can be only called when using SSI as DAI master
@@ -705,6 +731,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
+	if (!fsl_ssi_is_ac97(ssi_private)) {
+		u8 i2smode;
+		/*
+		 * Switch to normal net mode in order to have a frame sync
+		 * signal every 32 bits instead of 16 bits
+		 */
+		if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16)
+			i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL |
+				CCSR_SSI_SCR_NET;
+		else
+			i2smode = ssi_private->i2s_mode;
+
+		regmap_update_bits(regs, CCSR_SSI_SCR,
+				CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
+				channels == 1 ? 0 : i2smode);
+	}
+
 	/*
 	 * FIXME: The documentation says that SxCCR[WL] should not be
 	 * modified while the SSI is enabled.  The only time this can
@@ -724,11 +767,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 		regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK,
 				wl);
 
-	if (!fsl_ssi_is_ac97(ssi_private))
-		regmap_update_bits(regs, CCSR_SSI_SCR,
-				CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
-				channels == 1 ? 0 : ssi_private->i2s_mode);
-
 	return 0;
 }
 
@@ -781,6 +819,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBM_CFS:
 		case SND_SOC_DAIFMT_CBS_CFS:
 			ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER;
 			regmap_update_bits(regs, CCSR_SSI_STCCR,
@@ -854,6 +893,11 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 	case SND_SOC_DAIFMT_CBM_CFM:
 		scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
 		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		strcr &= ~CCSR_SSI_STCR_TXDIR;
+		strcr |= CCSR_SSI_STCR_TFDIR;
+		scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -1021,6 +1065,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
 
 static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
 	.startup	= fsl_ssi_startup,
+	.shutdown       = fsl_ssi_shutdown,
 	.hw_params	= fsl_ssi_hw_params,
 	.hw_free	= fsl_ssi_hw_free,
 	.set_fmt	= fsl_ssi_set_dai_fmt,
@@ -1146,17 +1191,22 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 	u32 dmas[4];
 	int ret;
 
-	ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
+	if (ssi_private->has_ipg_clk_name)
+		ssi_private->clk = devm_clk_get(&pdev->dev, "ipg");
+	else
+		ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
 	if (IS_ERR(ssi_private->clk)) {
 		ret = PTR_ERR(ssi_private->clk);
 		dev_err(&pdev->dev, "could not get clock: %d\n", ret);
 		return ret;
 	}
 
-	ret = clk_prepare_enable(ssi_private->clk);
-	if (ret) {
-		dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret);
-		return ret;
+	if (!ssi_private->has_ipg_clk_name) {
+		ret = clk_prepare_enable(ssi_private->clk);
+		if (ret) {
+			dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret);
+			return ret;
+		}
 	}
 
 	/* For those SLAVE implementations, we ingore non-baudclk cases
@@ -1214,8 +1264,9 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 	return 0;
 
 error_pcm:
-	clk_disable_unprepare(ssi_private->clk);
 
+	if (!ssi_private->has_ipg_clk_name)
+		clk_disable_unprepare(ssi_private->clk);
 	return ret;
 }
 
@@ -1224,7 +1275,8 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev,
 {
 	if (!ssi_private->use_dma)
 		imx_pcm_fiq_exit(pdev);
-	clk_disable_unprepare(ssi_private->clk);
+	if (!ssi_private->has_ipg_clk_name)
+		clk_disable_unprepare(ssi_private->clk);
 }
 
 static int fsl_ssi_probe(struct platform_device *pdev)
@@ -1263,9 +1315,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	if (sprop) {
 		if (!strcmp(sprop, "ac97-slave"))
 			ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97;
-		else if (!strcmp(sprop, "i2s-slave"))
-			ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S |
-				SND_SOC_DAIFMT_CBM_CFM;
 	}
 
 	ssi_private->use_dma = !of_property_read_bool(np,
@@ -1299,8 +1348,16 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	}
 
-	ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
+	ret = of_property_match_string(np, "clock-names", "ipg");
+	if (ret < 0) {
+		ssi_private->has_ipg_clk_name = false;
+		ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
 			&fsl_ssi_regconfig);
+	} else {
+		ssi_private->has_ipg_clk_name = true;
+		ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev,
+			"ipg", iomem, &fsl_ssi_regconfig);
+	}
 	if (IS_ERR(ssi_private->regs)) {
 		dev_err(&pdev->dev, "Failed to init register map\n");
 		return PTR_ERR(ssi_private->regs);
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
new file mode 100644
index 000000000000..f8cf10e16ce9
--- /dev/null
+++ b/sound/soc/fsl/imx-es8328.c
@@ -0,0 +1,233 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE	32
+#define MUX_PORT_MAX	7
+
+struct imx_es8328_data {
+	struct device *dev;
+	struct snd_soc_dai_link dai;
+	struct snd_soc_card card;
+	char codec_dai_name[DAI_NAME_SIZE];
+	char platform_name[DAI_NAME_SIZE];
+	int jack_gpio;
+};
+
+static struct snd_soc_jack_gpio headset_jack_gpios[] = {
+	{
+		.gpio = -1,
+		.name = "headset-gpio",
+		.report = SND_JACK_HEADSET,
+		.invert = 0,
+		.debounce_time = 200,
+	},
+};
+
+static struct snd_soc_jack headset_jack;
+
+static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct imx_es8328_data *data = container_of(rtd->card,
+					struct imx_es8328_data, card);
+	int ret = 0;
+
+	/* Headphone jack detection */
+	if (gpio_is_valid(data->jack_gpio)) {
+		ret = snd_soc_jack_new(rtd->codec, "Headphone",
+				       SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+				       &headset_jack);
+		if (ret)
+			return ret;
+
+		headset_jack_gpios[0].gpio = data->jack_gpio;
+		ret = snd_soc_jack_add_gpios(&headset_jack,
+					     ARRAY_SIZE(headset_jack_gpios),
+					     headset_jack_gpios);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0),
+};
+
+static int imx_es8328_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *ssi_np = NULL, *codec_np = NULL;
+	struct platform_device *ssi_pdev;
+	struct imx_es8328_data *data;
+	u32 int_port, ext_port;
+	int ret;
+	struct device *dev = &pdev->dev;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		goto fail;
+	}
+	if (int_port > MUX_PORT_MAX || int_port == 0) {
+		dev_err(dev, "mux-int-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		goto fail;
+	}
+
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		goto fail;
+	}
+	if (ext_port > MUX_PORT_MAX || ext_port == 0) {
+		dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the audmux API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+	ret = imx_audmux_v2_configure_port(int_port,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+	ret = imx_audmux_v2_configure_port(ext_port,
+			IMX_AUDMUX_V2_PTCR_SYN,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+	if (!ssi_np || !codec_np) {
+		dev_err(dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	ssi_pdev = of_find_device_by_node(ssi_np);
+	if (!ssi_pdev) {
+		dev_err(dev, "failed to find SSI platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto fail;
+	}
+
+	data->dev = dev;
+
+	data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+
+	data->dai.name = "hifi";
+	data->dai.stream_name = "hifi";
+	data->dai.codec_dai_name = "es8328-hifi-analog";
+	data->dai.codec_of_node = codec_np;
+	data->dai.cpu_of_node = ssi_np;
+	data->dai.platform_of_node = ssi_np;
+	data->dai.init = &imx_es8328_dai_init;
+	data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			    SND_SOC_DAIFMT_CBM_CFM;
+
+	data->card.dev = dev;
+	data->card.dapm_widgets = imx_es8328_dapm_widgets;
+	data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets);
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret) {
+		dev_err(dev, "Unable to parse card name\n");
+		goto fail;
+	}
+	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+	if (ret) {
+		dev_err(dev, "Unable to parse routing: %d\n", ret);
+		goto fail;
+	}
+	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
+	data->card.dai_link = &data->dai;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(dev, "Unable to register: %d\n", ret);
+		goto fail;
+	}
+
+	platform_set_drvdata(pdev, data);
+fail:
+	of_node_put(ssi_np);
+	of_node_put(codec_np);
+
+	return ret;
+}
+
+static int imx_es8328_remove(struct platform_device *pdev)
+{
+	struct imx_es8328_data *data = platform_get_drvdata(pdev);
+
+	snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios),
+				headset_jack_gpios);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_es8328_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-es8328", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids);
+
+static struct platform_driver imx_es8328_driver = {
+	.driver = {
+		.name = "imx-es8328",
+		.of_match_table = imx_es8328_dt_ids,
+	},
+	.probe = imx_es8328_probe,
+	.remove = imx_es8328_remove,
+};
+module_platform_driver(imx_es8328_driver);
+
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-audio-es8328");