Merge tag 'asoc-v5.2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v5.2

This is a pretty huge set of changes, it's been a pretty active release
all round but the big thing with this release is the Sound Open Firmware
changes from Intel, providing another DSP framework for use with the
DSPs in their SoCs.  This one works with the firmware of the same name
which is free software (unlike the previous DSP firmwares and framework)
and there has been some interest in adoption by other systems already so
hopefully we will see adoption by other vendors in the future.

Other highlights include:

 - Support for MCLK/sample rate ratio setting in the generic cards.
 - Support for pin switches in the generic cards.
 - A big set of improvements to the TLV320AIC32x4 drivers from Annaliese
   McDermond.
 - New drivers for Freescale audio mixers, several Intel machines,
   several Mediatek machines, Meson G12A, Sound Open Firmware and
   Spreadtrum compressed audio and DMA devices.

30 files changed, 1231 insertions, 286 deletions

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 7b1d9970be8b..55ed47c599e2 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI
 	  This option is only useful for out-of-tree drivers since
 	  in-tree drivers select it automatically.
 
+config SND_SOC_FSL_AUDMIX
+	tristate "Audio Mixer (AUDMIX) module support"
+	select REGMAP_MMIO
+	help
+	  Say Y if you want to add Audio Mixer (AUDMIX)
+	  support for the NXP iMX CPUs.
+
 config SND_SOC_FSL_SSI
 	tristate "Synchronous Serial Interface module (SSI) support"
 	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
@@ -182,16 +189,17 @@ config SND_MPC52xx_SOC_EFIKA
 
 endif # SND_POWERPC_SOC
 
+config SND_SOC_IMX_PCM_FIQ
+	tristate
+	default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC)
+	select FIQ
+
 if SND_IMX_SOC
 
 config SND_SOC_IMX_SSI
 	tristate
 	select SND_SOC_FSL_UTILS
 
-config SND_SOC_IMX_PCM_FIQ
-	tristate
-	select FIQ
-
 comment "SoC Audio support for Freescale i.MX boards:"
 
 config SND_MXC_SOC_WM1133_EV1
@@ -296,6 +304,15 @@ config SND_SOC_FSL_ASOC_CARD
 	 CS4271, CS4272 and SGTL5000.
 	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
 
+config SND_SOC_IMX_AUDMIX
+	tristate "SoC Audio support for i.MX boards with AUDMIX"
+	select SND_SOC_FSL_AUDMIX
+	select SND_SOC_FSL_SAI
+	help
+	  SoC Audio support for i.MX boards with Audio Mixer
+	  Say Y if you want to add support for SoC audio on an i.MX board with
+	  an Audio Mixer.
+
 endif # SND_IMX_SOC
 
 endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 3c0ff315b971..c0dd04422fe9 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -12,6 +12,7 @@ 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-audmix-objs := fsl_audmix.o
 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
@@ -22,6 +23,8 @@ snd-soc-fsl-esai-objs := fsl_esai.o
 snd-soc-fsl-micfil-objs := fsl_micfil.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+
+obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o
 obj-$(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
@@ -59,6 +62,7 @@ snd-soc-imx-es8328-objs := imx-es8328.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-spdif-objs := imx-spdif.o
 snd-soc-imx-mc13783-objs := imx-mc13783.o
+snd-soc-imx-audmix-objs := imx-audmix.o
 
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
@@ -68,3 +72,4 @@ obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
 obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
+obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 191426a6d9ad..d648268cb454 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -1,19 +1,13 @@
-/*
- * eukrea-tlv320.c  --  SoC audio for eukrea_cpuimxXX in I2S mode
- *
- * Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com>
- *
- * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
- * which is Copyright 2009 Simtec Electronics
- * and on sound/soc/imx/phycore-ac97.c which is
- * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
- *
- *  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;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// eukrea-tlv320.c  --  SoC audio for eukrea_cpuimxXX in I2S mode
+//
+// Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com>
+//
+// based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+// which is Copyright 2009 Simtec Electronics
+// and on sound/soc/imx/phycore-ac97.c which is
+// Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
 
 #include <linux/errno.h>
 #include <linux/module.h>
@@ -118,13 +112,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
 		if (ret) {
 			dev_err(&pdev->dev,
 				"fsl,mux-int-port node missing or invalid.\n");
-			return ret;
+			goto err;
 		}
 		ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port);
 		if (ret) {
 			dev_err(&pdev->dev,
 				"fsl,mux-ext-port node missing or invalid.\n");
-			return ret;
+			goto err;
 		}
 
 		/*
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
new file mode 100644
index 000000000000..3897a54a11fe
--- /dev/null
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -0,0 +1,578 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright 2017 NXP
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/pm_runtime.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_audmix.h"
+
+#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \
+	SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts)
+
+static const char
+	*tdm_sel[] = { "TDM1", "TDM2", },
+	*mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", },
+	*width_sel[] = { "16b", "18b", "20b", "24b", "32b", },
+	*endis_sel[] = { "Disabled", "Enabled", },
+	*updn_sel[] = { "Downward", "Upward", },
+	*mask_sel[] = { "Unmask", "Mask", };
+
+static const struct soc_enum fsl_audmix_enum[] = {
+/* FSL_AUDMIX_CTR enums */
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MIXCLK_SHIFT, tdm_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTSRC_SHIFT, mode_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTWIDTH_SHIFT, width_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKRTDF_SHIFT, mask_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKCKDF_SHIFT, mask_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCMODE_SHIFT, endis_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCSRC_SHIFT, tdm_sel),
+/* FSL_AUDMIX_ATCR0 enums */
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 0, endis_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 1, updn_sel),
+/* FSL_AUDMIX_ATCR1 enums */
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 0, endis_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 1, updn_sel),
+};
+
+struct fsl_audmix_state {
+	u8 tdms;
+	u8 clk;
+	char msg[64];
+};
+
+static const struct fsl_audmix_state prms[4][4] = {{
+	/* DIS->DIS, do nothing */
+	{ .tdms = 0, .clk = 0, .msg = "" },
+	/* DIS->TDM1*/
+	{ .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" },
+	/* DIS->TDM2*/
+	{ .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" },
+	/* DIS->MIX */
+	{ .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" }
+}, {	/* TDM1->DIS */
+	{ .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" },
+	/* TDM1->TDM1, do nothing */
+	{ .tdms = 0, .clk = 0, .msg = "" },
+	/* TDM1->TDM2 */
+	{ .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" },
+	/* TDM1->MIX */
+	{ .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" }
+}, {	/* TDM2->DIS */
+	{ .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" },
+	/* TDM2->TDM1 */
+	{ .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" },
+	/* TDM2->TDM2, do nothing */
+	{ .tdms = 0, .clk = 0, .msg = "" },
+	/* TDM2->MIX */
+	{ .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" }
+}, {	/* MIX->DIS */
+	{ .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" },
+	/* MIX->TDM1 */
+	{ .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" },
+	/* MIX->TDM2 */
+	{ .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" },
+	/* MIX->MIX, do nothing */
+	{ .tdms = 0, .clk = 0, .msg = "" }
+}, };
+
+static int fsl_audmix_state_trans(struct snd_soc_component *comp,
+				  unsigned int *mask, unsigned int *ctr,
+				  const struct fsl_audmix_state prm)
+{
+	struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp);
+	/* Enforce all required TDMs are started */
+	if ((priv->tdms & prm.tdms) != prm.tdms) {
+		dev_dbg(comp->dev, "%s", prm.msg);
+		return -EINVAL;
+	}
+
+	switch (prm.clk) {
+	case 1:
+	case 2:
+		/* Set mix clock */
+		(*mask) |= FSL_AUDMIX_CTR_MIXCLK_MASK;
+		(*ctr)  |= FSL_AUDMIX_CTR_MIXCLK(prm.clk - 1);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int fsl_audmix_put_mix_clk_src(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+	struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int *item = ucontrol->value.enumerated.item;
+	unsigned int reg_val, val, mix_clk;
+	int ret = 0;
+
+	/* Get current state */
+	ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, &reg_val);
+	if (ret)
+		return ret;
+
+	mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK)
+			>> FSL_AUDMIX_CTR_MIXCLK_SHIFT);
+	val = snd_soc_enum_item_to_val(e, item[0]);
+
+	dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val);
+
+	/**
+	 * Ensure the current selected mixer clock is available
+	 * for configuration propagation
+	 */
+	if (!(priv->tdms & BIT(mix_clk))) {
+		dev_err(comp->dev,
+			"Started TDM%d needed for config propagation!\n",
+			mix_clk + 1);
+		return -EINVAL;
+	}
+
+	if (!(priv->tdms & BIT(val))) {
+		dev_err(comp->dev,
+			"The selected clock source has no TDM%d enabled!\n",
+			val + 1);
+		return -EINVAL;
+	}
+
+	return snd_soc_put_enum_double(kcontrol, ucontrol);
+}
+
+static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+	struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int *item = ucontrol->value.enumerated.item;
+	u32 out_src, mix_clk;
+	unsigned int reg_val, val, mask = 0, ctr = 0;
+	int ret = 0;
+
+	/* Get current state */
+	ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, &reg_val);
+	if (ret)
+		return ret;
+
+	/* "From" state */
+	out_src = ((reg_val & FSL_AUDMIX_CTR_OUTSRC_MASK)
+			>> FSL_AUDMIX_CTR_OUTSRC_SHIFT);
+	mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK)
+			>> FSL_AUDMIX_CTR_MIXCLK_SHIFT);
+
+	/* "To" state */
+	val = snd_soc_enum_item_to_val(e, item[0]);
+
+	dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val);
+
+	/* Check if state is changing ... */
+	if (out_src == val)
+		return 0;
+	/**
+	 * Ensure the current selected mixer clock is available
+	 * for configuration propagation
+	 */
+	if (!(priv->tdms & BIT(mix_clk))) {
+		dev_err(comp->dev,
+			"Started TDM%d needed for config propagation!\n",
+			mix_clk + 1);
+		return -EINVAL;
+	}
+
+	/* Check state transition constraints */
+	ret = fsl_audmix_state_trans(comp, &mask, &ctr, prms[out_src][val]);
+	if (ret)
+		return ret;
+
+	/* Complete transition to new state */
+	mask |= FSL_AUDMIX_CTR_OUTSRC_MASK;
+	ctr  |= FSL_AUDMIX_CTR_OUTSRC(val);
+
+	return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr);
+}
+
+static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = {
+	/* FSL_AUDMIX_CTR controls */
+	SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0],
+		     snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src),
+	SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1],
+		     snd_soc_get_enum_double, fsl_audmix_put_out_src),
+	SOC_ENUM("Output Width", fsl_audmix_enum[2]),
+	SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]),
+	SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]),
+	SOC_ENUM("Sync Mode Config", fsl_audmix_enum[5]),
+	SOC_ENUM("Sync Mode Clk Source", fsl_audmix_enum[6]),
+	/* TDM1 Attenuation controls */
+	SOC_ENUM("TDM1 Attenuation", fsl_audmix_enum[7]),
+	SOC_ENUM("TDM1 Attenuation Direction", fsl_audmix_enum[8]),
+	SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AUDMIX_ATCR0,
+		   2, 0x00fff, 0),
+	SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AUDMIX_ATIVAL0,
+		   0, 0x3ffff, 0),
+	SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP0,
+		   0, 0x3ffff, 0),
+	SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN0,
+		   0, 0x3ffff, 0),
+	SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT0,
+		   0, 0x3ffff, 0),
+	/* TDM2 Attenuation controls */
+	SOC_ENUM("TDM2 Attenuation", fsl_audmix_enum[9]),
+	SOC_ENUM("TDM2 Attenuation Direction", fsl_audmix_enum[10]),
+	SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AUDMIX_ATCR1,
+		   2, 0x00fff, 0),
+	SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AUDMIX_ATIVAL1,
+		   0, 0x3ffff, 0),
+	SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP1,
+		   0, 0x3ffff, 0),
+	SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN1,
+		   0, 0x3ffff, 0),
+	SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT1,
+		   0, 0x3ffff, 0),
+};
+
+static int fsl_audmix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *comp = dai->component;
+	u32 mask = 0, ctr = 0;
+
+	/* AUDMIX is working in DSP_A format only */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* For playback the AUDMIX is slave, and for record is master */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_NF:
+		/* Output data will be written on positive edge of the clock */
+		ctr |= FSL_AUDMIX_CTR_OUTCKPOL(0);
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		/* Output data will be written on negative edge of the clock */
+		ctr |= FSL_AUDMIX_CTR_OUTCKPOL(1);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	mask |= FSL_AUDMIX_CTR_OUTCKPOL_MASK;
+
+	return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr);
+}
+
+static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *dai)
+{
+	struct fsl_audmix *priv = snd_soc_dai_get_drvdata(dai);
+
+	/* Capture stream shall not be handled */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		return 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		priv->tdms |= BIT(dai->driver->id);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		priv->tdms &= ~BIT(dai->driver->id);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops fsl_audmix_dai_ops = {
+	.set_fmt      = fsl_audmix_dai_set_fmt,
+	.trigger      = fsl_audmix_dai_trigger,
+};
+
+static struct snd_soc_dai_driver fsl_audmix_dai[] = {
+	{
+		.id   = 0,
+		.name = "audmix-0",
+		.playback = {
+			.stream_name = "AUDMIX-Playback-0",
+			.channels_min = 8,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 96000,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = FSL_AUDMIX_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AUDMIX-Capture-0",
+			.channels_min = 8,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 96000,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = FSL_AUDMIX_FORMATS,
+		},
+		.ops = &fsl_audmix_dai_ops,
+	},
+	{
+		.id   = 1,
+		.name = "audmix-1",
+		.playback = {
+			.stream_name = "AUDMIX-Playback-1",
+			.channels_min = 8,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 96000,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = FSL_AUDMIX_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AUDMIX-Capture-1",
+			.channels_min = 8,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 96000,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = FSL_AUDMIX_FORMATS,
+		},
+		.ops = &fsl_audmix_dai_ops,
+	},
+};
+
+static const struct snd_soc_component_driver fsl_audmix_component = {
+	.name		  = "fsl-audmix-dai",
+	.controls	  = fsl_audmix_snd_controls,
+	.num_controls	  = ARRAY_SIZE(fsl_audmix_snd_controls),
+};
+
+static bool fsl_audmix_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case FSL_AUDMIX_CTR:
+	case FSL_AUDMIX_STR:
+	case FSL_AUDMIX_ATCR0:
+	case FSL_AUDMIX_ATIVAL0:
+	case FSL_AUDMIX_ATSTPUP0:
+	case FSL_AUDMIX_ATSTPDN0:
+	case FSL_AUDMIX_ATSTPTGT0:
+	case FSL_AUDMIX_ATTNVAL0:
+	case FSL_AUDMIX_ATSTP0:
+	case FSL_AUDMIX_ATCR1:
+	case FSL_AUDMIX_ATIVAL1:
+	case FSL_AUDMIX_ATSTPUP1:
+	case FSL_AUDMIX_ATSTPDN1:
+	case FSL_AUDMIX_ATSTPTGT1:
+	case FSL_AUDMIX_ATTNVAL1:
+	case FSL_AUDMIX_ATSTP1:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool fsl_audmix_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case FSL_AUDMIX_CTR:
+	case FSL_AUDMIX_ATCR0:
+	case FSL_AUDMIX_ATIVAL0:
+	case FSL_AUDMIX_ATSTPUP0:
+	case FSL_AUDMIX_ATSTPDN0:
+	case FSL_AUDMIX_ATSTPTGT0:
+	case FSL_AUDMIX_ATCR1:
+	case FSL_AUDMIX_ATIVAL1:
+	case FSL_AUDMIX_ATSTPUP1:
+	case FSL_AUDMIX_ATSTPDN1:
+	case FSL_AUDMIX_ATSTPTGT1:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const struct reg_default fsl_audmix_reg[] = {
+	{ FSL_AUDMIX_CTR,       0x00060 },
+	{ FSL_AUDMIX_STR,       0x00003 },
+	{ FSL_AUDMIX_ATCR0,     0x00000 },
+	{ FSL_AUDMIX_ATIVAL0,   0x3FFFF },
+	{ FSL_AUDMIX_ATSTPUP0,  0x2AAAA },
+	{ FSL_AUDMIX_ATSTPDN0,  0x30000 },
+	{ FSL_AUDMIX_ATSTPTGT0, 0x00010 },
+	{ FSL_AUDMIX_ATTNVAL0,  0x00000 },
+	{ FSL_AUDMIX_ATSTP0,    0x00000 },
+	{ FSL_AUDMIX_ATCR1,     0x00000 },
+	{ FSL_AUDMIX_ATIVAL1,   0x3FFFF },
+	{ FSL_AUDMIX_ATSTPUP1,  0x2AAAA },
+	{ FSL_AUDMIX_ATSTPDN1,  0x30000 },
+	{ FSL_AUDMIX_ATSTPTGT1, 0x00010 },
+	{ FSL_AUDMIX_ATTNVAL1,  0x00000 },
+	{ FSL_AUDMIX_ATSTP1,    0x00000 },
+};
+
+static const struct regmap_config fsl_audmix_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = FSL_AUDMIX_ATSTP1,
+	.reg_defaults = fsl_audmix_reg,
+	.num_reg_defaults = ARRAY_SIZE(fsl_audmix_reg),
+	.readable_reg = fsl_audmix_readable_reg,
+	.writeable_reg = fsl_audmix_writeable_reg,
+	.cache_type = REGCACHE_FLAT,
+};
+
+static const struct of_device_id fsl_audmix_ids[] = {
+	{
+		.compatible = "fsl,imx8qm-audmix",
+		.data = "imx-audmix",
+	},
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_audmix_ids);
+
+static int fsl_audmix_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct fsl_audmix *priv;
+	struct resource *res;
+	const char *mdrv;
+	const struct of_device_id *of_id;
+	void __iomem *regs;
+	int ret;
+
+	of_id = of_match_device(fsl_audmix_ids, dev);
+	if (!of_id || !of_id->data)
+		return -EINVAL;
+
+	mdrv = of_id->data;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	/* Get the addresses */
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	priv->regmap = devm_regmap_init_mmio_clk(dev, "ipg", regs,
+						 &fsl_audmix_regmap_config);
+	if (IS_ERR(priv->regmap)) {
+		dev_err(dev, "failed to init regmap\n");
+		return PTR_ERR(priv->regmap);
+	}
+
+	priv->ipg_clk = devm_clk_get(dev, "ipg");
+	if (IS_ERR(priv->ipg_clk)) {
+		dev_err(dev, "failed to get ipg clock\n");
+		return PTR_ERR(priv->ipg_clk);
+	}
+
+	platform_set_drvdata(pdev, priv);
+	pm_runtime_enable(dev);
+
+	ret = devm_snd_soc_register_component(dev, &fsl_audmix_component,
+					      fsl_audmix_dai,
+					      ARRAY_SIZE(fsl_audmix_dai));
+	if (ret) {
+		dev_err(dev, "failed to register ASoC DAI\n");
+		return ret;
+	}
+
+	priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0);
+	if (IS_ERR(priv->pdev)) {
+		ret = PTR_ERR(priv->pdev);
+		dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret);
+	}
+
+	return ret;
+}
+
+static int fsl_audmix_remove(struct platform_device *pdev)
+{
+	struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev);
+
+	if (priv->pdev)
+		platform_device_unregister(priv->pdev);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int fsl_audmix_runtime_resume(struct device *dev)
+{
+	struct fsl_audmix *priv = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_prepare_enable(priv->ipg_clk);
+	if (ret) {
+		dev_err(dev, "Failed to enable IPG clock: %d\n", ret);
+		return ret;
+	}
+
+	regcache_cache_only(priv->regmap, false);
+	regcache_mark_dirty(priv->regmap);
+
+	return regcache_sync(priv->regmap);
+}
+
+static int fsl_audmix_runtime_suspend(struct device *dev)
+{
+	struct fsl_audmix *priv = dev_get_drvdata(dev);
+
+	regcache_cache_only(priv->regmap, true);
+
+	clk_disable_unprepare(priv->ipg_clk);
+
+	return 0;
+}
+#endif /* CONFIG_PM */
+
+static const struct dev_pm_ops fsl_audmix_pm = {
+	SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend,
+			   fsl_audmix_runtime_resume,
+			   NULL)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
+};
+
+static struct platform_driver fsl_audmix_driver = {
+	.probe = fsl_audmix_probe,
+	.remove = fsl_audmix_remove,
+	.driver = {
+		.name = "fsl-audmix",
+		.of_match_table = fsl_audmix_ids,
+		.pm = &fsl_audmix_pm,
+	},
+};
+module_platform_driver(fsl_audmix_driver);
+
+MODULE_DESCRIPTION("NXP AUDMIX ASoC DAI driver");
+MODULE_AUTHOR("Viorel Suman <viorel.suman@nxp.com>");
+MODULE_ALIAS("platform:fsl-audmix");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_audmix.h b/sound/soc/fsl/fsl_audmix.h
new file mode 100644
index 000000000000..7812ffec45c5
--- /dev/null
+++ b/sound/soc/fsl/fsl_audmix.h
@@ -0,0 +1,102 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright 2017 NXP
+ */
+
+#ifndef __FSL_AUDMIX_H
+#define __FSL_AUDMIX_H
+
+#define FSL_AUDMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			SNDRV_PCM_FMTBIT_S24_LE |\
+			SNDRV_PCM_FMTBIT_S32_LE)
+/* AUDMIX Registers */
+#define FSL_AUDMIX_CTR		0x200 /* Control */
+#define FSL_AUDMIX_STR		0x204 /* Status */
+
+#define FSL_AUDMIX_ATCR0	0x208 /* Attenuation Control */
+#define FSL_AUDMIX_ATIVAL0	0x20c /* Attenuation Initial Value */
+#define FSL_AUDMIX_ATSTPUP0	0x210 /* Attenuation step up factor */
+#define FSL_AUDMIX_ATSTPDN0	0x214 /* Attenuation step down factor */
+#define FSL_AUDMIX_ATSTPTGT0	0x218 /* Attenuation step target */
+#define FSL_AUDMIX_ATTNVAL0	0x21c /* Attenuation Value */
+#define FSL_AUDMIX_ATSTP0	0x220 /* Attenuation step number */
+
+#define FSL_AUDMIX_ATCR1	0x228 /* Attenuation Control */
+#define FSL_AUDMIX_ATIVAL1	0x22c /* Attenuation Initial Value */
+#define FSL_AUDMIX_ATSTPUP1	0x230 /* Attenuation step up factor */
+#define FSL_AUDMIX_ATSTPDN1	0x234 /* Attenuation step down factor */
+#define FSL_AUDMIX_ATSTPTGT1	0x238 /* Attenuation step target */
+#define FSL_AUDMIX_ATTNVAL1	0x23c /* Attenuation Value */
+#define FSL_AUDMIX_ATSTP1	0x240 /* Attenuation step number */
+
+/* AUDMIX Control Register */
+#define FSL_AUDMIX_CTR_MIXCLK_SHIFT	0
+#define FSL_AUDMIX_CTR_MIXCLK_MASK	BIT(FSL_AUDMIX_CTR_MIXCLK_SHIFT)
+#define FSL_AUDMIX_CTR_MIXCLK(i)	((i) << FSL_AUDMIX_CTR_MIXCLK_SHIFT)
+#define FSL_AUDMIX_CTR_OUTSRC_SHIFT	1
+#define FSL_AUDMIX_CTR_OUTSRC_MASK	(0x3 << FSL_AUDMIX_CTR_OUTSRC_SHIFT)
+#define FSL_AUDMIX_CTR_OUTSRC(i)	(((i) << FSL_AUDMIX_CTR_OUTSRC_SHIFT)\
+					      & FSL_AUDMIX_CTR_OUTSRC_MASK)
+#define FSL_AUDMIX_CTR_OUTWIDTH_SHIFT	3
+#define FSL_AUDMIX_CTR_OUTWIDTH_MASK	(0x7 << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT)
+#define FSL_AUDMIX_CTR_OUTWIDTH(i)	(((i) << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT)\
+					      & FSL_AUDMIX_CTR_OUTWIDTH_MASK)
+#define FSL_AUDMIX_CTR_OUTCKPOL_SHIFT	6
+#define FSL_AUDMIX_CTR_OUTCKPOL_MASK	BIT(FSL_AUDMIX_CTR_OUTCKPOL_SHIFT)
+#define FSL_AUDMIX_CTR_OUTCKPOL(i)	((i) << FSL_AUDMIX_CTR_OUTCKPOL_SHIFT)
+#define FSL_AUDMIX_CTR_MASKRTDF_SHIFT	7
+#define FSL_AUDMIX_CTR_MASKRTDF_MASK	BIT(FSL_AUDMIX_CTR_MASKRTDF_SHIFT)
+#define FSL_AUDMIX_CTR_MASKRTDF(i)	((i) << FSL_AUDMIX_CTR_MASKRTDF_SHIFT)
+#define FSL_AUDMIX_CTR_MASKCKDF_SHIFT	8
+#define FSL_AUDMIX_CTR_MASKCKDF_MASK	BIT(FSL_AUDMIX_CTR_MASKCKDF_SHIFT)
+#define FSL_AUDMIX_CTR_MASKCKDF(i)	((i) << FSL_AUDMIX_CTR_MASKCKDF_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCMODE_SHIFT	9
+#define FSL_AUDMIX_CTR_SYNCMODE_MASK	BIT(FSL_AUDMIX_CTR_SYNCMODE_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCMODE(i)	((i) << FSL_AUDMIX_CTR_SYNCMODE_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCSRC_SHIFT	10
+#define FSL_AUDMIX_CTR_SYNCSRC_MASK	BIT(FSL_AUDMIX_CTR_SYNCSRC_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCSRC(i)	((i) << FSL_AUDMIX_CTR_SYNCSRC_SHIFT)
+
+/* AUDMIX Status Register */
+#define FSL_AUDMIX_STR_RATEDIFF		BIT(0)
+#define FSL_AUDMIX_STR_CLKDIFF		BIT(1)
+#define FSL_AUDMIX_STR_MIXSTAT_SHIFT	2
+#define FSL_AUDMIX_STR_MIXSTAT_MASK	(0x3 << FSL_AUDMIX_STR_MIXSTAT_SHIFT)
+#define FSL_AUDMIX_STR_MIXSTAT(i)	(((i) & FSL_AUDMIX_STR_MIXSTAT_MASK) \
+					   >> FSL_AUDMIX_STR_MIXSTAT_SHIFT)
+/* AUDMIX Attenuation Control Register */
+#define FSL_AUDMIX_ATCR_AT_EN		BIT(0)
+#define FSL_AUDMIX_ATCR_AT_UPDN		BIT(1)
+#define FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT	2
+#define FSL_AUDMIX_ATCR_ATSTPDFI_MASK	\
+				(0xfff << FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT)
+
+/* AUDMIX Attenuation Initial Value Register */
+#define FSL_AUDMIX_ATIVAL_ATINVAL_MASK	0x3FFFF
+
+/* AUDMIX Attenuation Step Up Factor Register */
+#define FSL_AUDMIX_ATSTPUP_ATSTEPUP_MASK	0x3FFFF
+
+/* AUDMIX Attenuation Step Down Factor Register */
+#define FSL_AUDMIX_ATSTPDN_ATSTEPDN_MASK	0x3FFFF
+
+/* AUDMIX Attenuation Step Target Register */
+#define FSL_AUDMIX_ATSTPTGT_ATSTPTG_MASK	0x3FFFF
+
+/* AUDMIX Attenuation Value Register */
+#define FSL_AUDMIX_ATTNVAL_ATCURVAL_MASK	0x3FFFF
+
+/* AUDMIX Attenuation Step Number Register */
+#define FSL_AUDMIX_ATSTP_STPCTR_MASK	0x3FFFF
+
+#define FSL_AUDMIX_MAX_DAIS		2
+struct fsl_audmix {
+	struct platform_device *pdev;
+	struct regmap *regmap;
+	struct clk *ipg_clk;
+	u8 tdms;
+};
+
+#endif /* __FSL_AUDMIX_H */
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 78871de35086..e22508301412 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -1,18 +1,14 @@
-/*
- * Freescale DMA ALSA SoC PCM driver
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2007-2010 Freescale Semiconductor, Inc.
- *
- * 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.
- *
- * This driver implements ASoC support for the Elo DMA controller, which is
- * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms,
- * the PCM driver is what handles the DMA buffer.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale DMA ALSA SoC PCM driver
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2007-2010 Freescale Semiconductor, Inc.
+//
+// This driver implements ASoC support for the Elo DMA controller, which is
+// the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms,
+// the PCM driver is what handles the DMA buffer.
 
 #include <linux/module.h>
 #include <linux/init.h>
diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h
index 78fee97e8036..f19ae765b656 100644
--- a/sound/soc/fsl/fsl_dma.h
+++ b/sound/soc/fsl/fsl_dma.h
@@ -1,9 +1,6 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef _MPC8610_PCM_H
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 3623aa9a6f2e..bad0dfed6b68 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -218,7 +218,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 {
 	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
 	struct clk *clksrc = esai_priv->extalclk;
-	bool tx = clk_id <= ESAI_HCKT_EXTAL;
+	bool tx = (clk_id <= ESAI_HCKT_EXTAL || esai_priv->synchronous);
 	bool in = dir == SND_SOC_CLOCK_IN;
 	u32 ratio, ecr = 0;
 	unsigned long clk_rate;
@@ -251,9 +251,9 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 		break;
 	case ESAI_HCKT_EXTAL:
 		ecr |= ESAI_ECR_ETI;
-		/* fall through */
+		break;
 	case ESAI_HCKR_EXTAL:
-		ecr |= ESAI_ECR_ERI;
+		ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI;
 		break;
 	default:
 		return -EINVAL;
@@ -537,10 +537,18 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 
 	bclk = params_rate(params) * slot_width * esai_priv->slots;
 
-	ret = fsl_esai_set_bclk(dai, tx, bclk);
+	ret = fsl_esai_set_bclk(dai, esai_priv->synchronous || tx, bclk);
 	if (ret)
 		return ret;
 
+	mask = ESAI_xCR_xSWS_MASK;
+	val = ESAI_xCR_xSWS(slot_width, width);
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
+	/* Recording in synchronous mode needs to set TCR also */
+	if (!tx && esai_priv->synchronous)
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, val);
+
 	/* Use Normal mode to support monaural audio */
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 			   ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ?
@@ -556,10 +564,9 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
 
-	mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0);
-	val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
-
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
+	if (tx)
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR,
+				ESAI_xCR_PADC, ESAI_xCR_PADC);
 
 	/* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 40c07e756481..f7f2d29f1bfe 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -151,12 +151,9 @@ static inline int get_clk_div(struct fsl_micfil *micfil,
 {
 	u32 ctrl2_reg;
 	long mclk_rate;
-	int osr;
 	int clk_div;
 
 	regmap_read(micfil->regmap, REG_MICFIL_CTRL2, &ctrl2_reg);
-	osr = 16 - ((ctrl2_reg & MICFIL_CTRL2_CICOSR_MASK)
-		    >> MICFIL_CTRL2_CICOSR_SHIFT);
 
 	mclk_rate = clk_get_rate(micfil->mclk);
 
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index db9e0872f73d..8593269156bd 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -9,6 +9,7 @@
 #include <linux/dmaengine.h>
 #include <linux/module.h>
 #include <linux/of_address.h>
+#include <linux/pm_runtime.h>
 #include <linux/regmap.h>
 #include <linux/slab.h>
 #include <linux/time.h>
@@ -268,12 +269,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_CBS_CFS:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+		sai->is_slave_mode = false;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		sai->is_slave_mode = true;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFM:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+		sai->is_slave_mode = false;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
@@ -899,6 +902,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(pdev, sai);
 
+	pm_runtime_enable(&pdev->dev);
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
 			&fsl_sai_dai, 1);
 	if (ret)
@@ -910,6 +915,13 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 }
 
+static int fsl_sai_remove(struct platform_device *pdev)
+{
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
 static const struct of_device_id fsl_sai_ids[] = {
 	{ .compatible = "fsl,vf610-sai", },
 	{ .compatible = "fsl,imx6sx-sai", },
@@ -918,8 +930,8 @@ static const struct of_device_id fsl_sai_ids[] = {
 };
 MODULE_DEVICE_TABLE(of, fsl_sai_ids);
 
-#ifdef CONFIG_PM_SLEEP
-static int fsl_sai_suspend(struct device *dev)
+#ifdef CONFIG_PM
+static int fsl_sai_runtime_suspend(struct device *dev)
 {
 	struct fsl_sai *sai = dev_get_drvdata(dev);
 
@@ -929,7 +941,7 @@ static int fsl_sai_suspend(struct device *dev)
 	return 0;
 }
 
-static int fsl_sai_resume(struct device *dev)
+static int fsl_sai_runtime_resume(struct device *dev)
 {
 	struct fsl_sai *sai = dev_get_drvdata(dev);
 
@@ -941,14 +953,18 @@ static int fsl_sai_resume(struct device *dev)
 	regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
 	return regcache_sync(sai->regmap);
 }
-#endif /* CONFIG_PM_SLEEP */
+#endif /* CONFIG_PM */
 
 static const struct dev_pm_ops fsl_sai_pm_ops = {
-	SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume)
+	SET_RUNTIME_PM_OPS(fsl_sai_runtime_suspend,
+			   fsl_sai_runtime_resume, NULL)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
 };
 
 static struct platform_driver fsl_sai_driver = {
 	.probe = fsl_sai_probe,
+	.remove = fsl_sai_remove,
 	.driver = {
 		.name = "fsl-sai",
 		.pm = &fsl_sai_pm_ops,
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 9981668ab590..040d06b89f00 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -71,6 +71,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
 	iprop = of_get_property(dma_np, "cell-index", NULL);
 	if (!iprop) {
 		of_node_put(dma_np);
+		of_node_put(dma_channel_np);
 		return -EINVAL;
 	}
 	*dma_id = be32_to_cpup(iprop);
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
new file mode 100644
index 000000000000..9aaf3e5b45b9
--- /dev/null
+++ b/sound/soc/fsl/imx-audmix.c
@@ -0,0 +1,331 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright 2017 NXP
+ *
+ * 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/module.h>
+#include <linux/of_platform.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <linux/pm_runtime.h>
+#include "fsl_sai.h"
+#include "fsl_audmix.h"
+
+struct imx_audmix {
+	struct platform_device *pdev;
+	struct snd_soc_card card;
+	struct platform_device *audmix_pdev;
+	struct platform_device *out_pdev;
+	struct clk *cpu_mclk;
+	int num_dai;
+	struct snd_soc_dai_link *dai;
+	int num_dai_conf;
+	struct snd_soc_codec_conf *dai_conf;
+	int num_dapm_routes;
+	struct snd_soc_dapm_route *dapm_routes;
+};
+
+static const u32 imx_audmix_rates[] = {
+	8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000,
+};
+
+static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = {
+	.count = ARRAY_SIZE(imx_audmix_rates),
+	.list = imx_audmix_rates,
+};
+
+static int imx_audmix_fe_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct device *dev = rtd->card->dev;
+	unsigned long clk_rate = clk_get_rate(priv->cpu_mclk);
+	int ret;
+
+	if (clk_rate % 24576000 == 0) {
+		ret = snd_pcm_hw_constraint_list(runtime, 0,
+						 SNDRV_PCM_HW_PARAM_RATE,
+						 &imx_audmix_rate_constraints);
+		if (ret < 0)
+			return ret;
+	} else {
+		dev_warn(dev, "mclk may be not supported %lu\n", clk_rate);
+	}
+
+	ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS,
+					   1, 8);
+	if (ret < 0)
+		return ret;
+
+	return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT,
+					    FSL_AUDMIX_FORMATS);
+}
+
+static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->card->dev;
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF;
+	u32 channels = params_channels(params);
+	int ret, dir;
+
+	/* For playback the AUDMIX is slave, and for record is master */
+	fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM;
+	dir  = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN;
+
+	/* set DAI configuration */
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+	if (ret) {
+		dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir);
+	if (ret) {
+		dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * Per datasheet, AUDMIX expects 8 slots and 32 bits
+	 * for every slot in TDM mode.
+	 */
+	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1,
+				       BIT(channels) - 1, 8, 32);
+	if (ret)
+		dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret);
+
+	return ret;
+}
+
+static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->card->dev;
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF;
+	int ret;
+
+	if (!tx)
+		return 0;
+
+	/* For playback the AUDMIX is slave */
+	fmt |= SND_SOC_DAIFMT_CBM_CFM;
+
+	/* set AUDMIX DAI configuration */
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+	if (ret)
+		dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret);
+
+	return ret;
+}
+
+static struct snd_soc_ops imx_audmix_fe_ops = {
+	.startup = imx_audmix_fe_startup,
+	.hw_params = imx_audmix_fe_hw_params,
+};
+
+static struct snd_soc_ops imx_audmix_be_ops = {
+	.hw_params = imx_audmix_be_hw_params,
+};
+
+static int imx_audmix_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *audmix_np = NULL, *out_cpu_np = NULL;
+	struct platform_device *audmix_pdev = NULL;
+	struct platform_device *cpu_pdev;
+	struct of_phandle_args args;
+	struct imx_audmix *priv;
+	int i, num_dai, ret;
+	const char *fe_name_pref = "HiFi-AUDMIX-FE-";
+	char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name;
+
+	if (pdev->dev.parent) {
+		audmix_np = pdev->dev.parent->of_node;
+	} else {
+		dev_err(&pdev->dev, "Missing parent device.\n");
+		return -EINVAL;
+	}
+
+	if (!audmix_np) {
+		dev_err(&pdev->dev, "Missing DT node for parent device.\n");
+		return -EINVAL;
+	}
+
+	audmix_pdev = of_find_device_by_node(audmix_np);
+	if (!audmix_pdev) {
+		dev_err(&pdev->dev, "Missing AUDMIX platform device for %s\n",
+			np->full_name);
+		return -EINVAL;
+	}
+	put_device(&audmix_pdev->dev);
+
+	num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL);
+	if (num_dai != FSL_AUDMIX_MAX_DAIS) {
+		dev_err(&pdev->dev, "Need 2 dais to be provided for %s\n",
+			audmix_np->full_name);
+		return -EINVAL;
+	}
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->num_dai = 2 * num_dai;
+	priv->dai = devm_kzalloc(&pdev->dev, priv->num_dai *
+				 sizeof(struct snd_soc_dai_link), GFP_KERNEL);
+	if (!priv->dai)
+		return -ENOMEM;
+
+	priv->num_dai_conf = num_dai;
+	priv->dai_conf = devm_kzalloc(&pdev->dev, priv->num_dai_conf *
+				      sizeof(struct snd_soc_codec_conf),
+				      GFP_KERNEL);
+	if (!priv->dai_conf)
+		return -ENOMEM;
+
+	priv->num_dapm_routes = 3 * num_dai;
+	priv->dapm_routes = devm_kzalloc(&pdev->dev, priv->num_dapm_routes *
+					 sizeof(struct snd_soc_dapm_route),
+					 GFP_KERNEL);
+	if (!priv->dapm_routes)
+		return -ENOMEM;
+
+	for (i = 0; i < num_dai; i++) {
+		ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i,
+						 &args);
+		if (ret < 0) {
+			dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n");
+			return ret;
+		}
+
+		cpu_pdev = of_find_device_by_node(args.np);
+		if (!cpu_pdev) {
+			dev_err(&pdev->dev, "failed to find SAI platform device\n");
+			return -EINVAL;
+		}
+		put_device(&cpu_pdev->dev);
+
+		dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s",
+					  fe_name_pref, args.np->full_name + 1);
+
+		dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name);
+
+		if (i == 0) {
+			out_cpu_np = args.np;
+			capture_dai_name =
+				devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
+					       dai_name, "CPU-Capture");
+		}
+
+		priv->dai[i].name = dai_name;
+		priv->dai[i].stream_name = "HiFi-AUDMIX-FE";
+		priv->dai[i].codec_dai_name = "snd-soc-dummy-dai";
+		priv->dai[i].codec_name = "snd-soc-dummy";
+		priv->dai[i].cpu_of_node = args.np;
+		priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev);
+		priv->dai[i].platform_of_node = args.np;
+		priv->dai[i].dynamic = 1;
+		priv->dai[i].dpcm_playback = 1;
+		priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0);
+		priv->dai[i].ignore_pmdown_time = 1;
+		priv->dai[i].ops = &imx_audmix_fe_ops;
+
+		/* Add AUDMIX Backend */
+		be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL,
+					 "audmix-%d", i);
+		be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL,
+				       "AUDMIX-Playback-%d", i);
+		be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL,
+				       "AUDMIX-Capture-%d", i);
+
+		priv->dai[num_dai + i].name = be_name;
+		priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai";
+		priv->dai[num_dai + i].codec_name = "snd-soc-dummy";
+		priv->dai[num_dai + i].cpu_of_node = audmix_np;
+		priv->dai[num_dai + i].cpu_dai_name = be_name;
+		priv->dai[num_dai + i].platform_name = "snd-soc-dummy";
+		priv->dai[num_dai + i].no_pcm = 1;
+		priv->dai[num_dai + i].dpcm_playback = 1;
+		priv->dai[num_dai + i].dpcm_capture  = 1;
+		priv->dai[num_dai + i].ignore_pmdown_time = 1;
+		priv->dai[num_dai + i].ops = &imx_audmix_be_ops;
+
+		priv->dai_conf[i].of_node = args.np;
+		priv->dai_conf[i].name_prefix = dai_name;
+
+		priv->dapm_routes[i].source =
+			devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
+				       dai_name, "CPU-Playback");
+		priv->dapm_routes[i].sink = be_pb;
+		priv->dapm_routes[num_dai + i].source   = be_pb;
+		priv->dapm_routes[num_dai + i].sink     = be_cp;
+		priv->dapm_routes[2 * num_dai + i].source = be_cp;
+		priv->dapm_routes[2 * num_dai + i].sink   = capture_dai_name;
+	}
+
+	cpu_pdev = of_find_device_by_node(out_cpu_np);
+	if (!cpu_pdev) {
+		dev_err(&pdev->dev, "failed to find SAI platform device\n");
+		return -EINVAL;
+	}
+	put_device(&cpu_pdev->dev);
+
+	priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1");
+	if (IS_ERR(priv->cpu_mclk)) {
+		ret = PTR_ERR(priv->cpu_mclk);
+		dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret);
+		return -EINVAL;
+	}
+
+	priv->audmix_pdev = audmix_pdev;
+	priv->out_pdev  = cpu_pdev;
+
+	priv->card.dai_link = priv->dai;
+	priv->card.num_links = priv->num_dai;
+	priv->card.codec_conf = priv->dai_conf;
+	priv->card.num_configs = priv->num_dai_conf;
+	priv->card.dapm_routes = priv->dapm_routes;
+	priv->card.num_dapm_routes = priv->num_dapm_routes;
+	priv->card.dev = pdev->dev.parent;
+	priv->card.owner = THIS_MODULE;
+	priv->card.name = "imx-audmix";
+
+	platform_set_drvdata(pdev, &priv->card);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct platform_driver imx_audmix_driver = {
+	.probe = imx_audmix_probe,
+	.driver = {
+		.name = "imx-audmix",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+module_platform_driver(imx_audmix_driver);
+
+MODULE_DESCRIPTION("NXP AUDMIX ASoC machine driver");
+MODULE_AUTHOR("Viorel Suman <viorel.suman@nxp.com>");
+MODULE_ALIAS("platform:imx-audmix");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 99e07b01a2ce..04e59e66711d 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -1,21 +1,11 @@
-/*
- * Copyright 2012 Freescale Semiconductor, Inc.
- * Copyright 2012 Linaro Ltd.
- * Copyright 2009 Pengutronix, Sascha Hauer <s.hauer@pengutronix.de>
- *
- * Initial development of this code was funded by
- * Phytec Messtechnik GmbH, http://www.phytec.de
- *
- * 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; either version 2 of the License, or
- * (at your option) any later version.
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright 2012 Freescale Semiconductor, Inc.
+// Copyright 2012 Linaro Ltd.
+// Copyright 2009 Pengutronix, Sascha Hauer <s.hauer@pengutronix.de>
+//
+// Initial development of this code was funded by
+// Phytec Messtechnik GmbH, http://www.phytec.de
 
 #include <linux/clk.h>
 #include <linux/debugfs.h>
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index 9953438086e4..c9d8739b04a9 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -1,14 +1,7 @@
-/*
- * 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
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright 2012 Freescale Semiconductor, Inc.
+// Copyright 2012 Linaro Ltd.
 
 #include <linux/gpio.h>
 #include <linux/module.h>
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 9d19b808f634..545815a27074 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -1,17 +1,11 @@
-/*
- * imx-mc13783.c  --  SoC audio for imx based boards with mc13783 codec
- *
- * Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch>
- *
- * Heavly based on phycore-mc13783:
- * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
- *
- *  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;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// imx-mc13783.c  --  SoC audio for imx based boards with mc13783 codec
+//
+// Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch>
+//
+// Heavly based on phycore-mc13783:
+// Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 0578f3486847..c49aea4fba56 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -1,16 +1,11 @@
-/*
- * imx-pcm-fiq.c  --  ALSA Soc Audio Layer
- *
- * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
- *
- * This code is based on code copyrighted by Freescale,
- * Liam Girdwood, Javier Martin and probably others.
- *
- *  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;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+// imx-pcm-fiq.c  --  ALSA Soc Audio Layer
+//
+// Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
+//
+// This code is based on code copyrighted by Freescale,
+// Liam Girdwood, Javier Martin and probably others.
+
 #include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/device.h>
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index 133c4470acad..5dd406774d3e 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -1,13 +1,9 @@
+/* SPDX-License-Identifier: GPL-2.0+ */
 /*
  * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
  *
  * This code is based on code copyrighted by Freescale,
  * Liam Girdwood, Javier Martin and probably others.
- *
- * 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;  either version 2 of the  License, or (at your
- * option) any later version.
  */
 
 #ifndef _IMX_PCM_H
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 797d66e43d49..4f7f210beb18 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -1,13 +1,6 @@
-/*
- * Copyright (C) 2013 Freescale Semiconductor, Inc.
- *
- * 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
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright (C) 2013 Freescale Semiconductor, Inc.
 
 #include <linux/module.h>
 #include <linux/of_platform.h>
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 06790615e04e..9038b61317be 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -1,35 +1,28 @@
-/*
- * imx-ssi.c  --  ALSA Soc Audio Layer
- *
- * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
- *
- * This code is based on code copyrighted by Freescale,
- * Liam Girdwood, Javier Martin and probably others.
- *
- *  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;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- *
- * The i.MX SSI core has some nasty limitations in AC97 mode. While most
- * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
- * one FIFO which combines all valid receive slots. We cannot even select
- * which slots we want to receive. The WM9712 with which this driver
- * was developed with always sends GPIO status data in slot 12 which
- * we receive in our (PCM-) data stream. The only chance we have is to
- * manually skip this data in the FIQ handler. With sampling rates different
- * from 48000Hz not every frame has valid receive data, so the ratio
- * between pcm data and GPIO status data changes. Our FIQ handler is not
- * able to handle this, hence this driver only works with 48000Hz sampling
- * rate.
- * Reading and writing AC97 registers is another challenge. The core
- * provides us status bits when the read register is updated with *another*
- * value. When we read the same register two times (and the register still
- * contains the same value) these status bits are not set. We work
- * around this by not polling these bits but only wait a fixed delay.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// imx-ssi.c  --  ALSA Soc Audio Layer
+//
+// Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
+//
+// This code is based on code copyrighted by Freescale,
+// Liam Girdwood, Javier Martin and probably others.
+//
+// The i.MX SSI core has some nasty limitations in AC97 mode. While most
+// sane processor vendors have a FIFO per AC97 slot, the i.MX has only
+// one FIFO which combines all valid receive slots. We cannot even select
+// which slots we want to receive. The WM9712 with which this driver
+// was developed with always sends GPIO status data in slot 12 which
+// we receive in our (PCM-) data stream. The only chance we have is to
+// manually skip this data in the FIQ handler. With sampling rates different
+// from 48000Hz not every frame has valid receive data, so the ratio
+// between pcm data and GPIO status data changes. Our FIQ handler is not
+// able to handle this, hence this driver only works with 48000Hz sampling
+// rate.
+// Reading and writing AC97 registers is another challenge. The core
+// provides us status bits when the read register is updated with *another*
+// value. When we read the same register two times (and the register still
+// contains the same value) these status bits are not set. We work
+// around this by not polling these bits but only wait a fixed delay.
 
 #include <linux/clk.h>
 #include <linux/delay.h>
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index be6562365b6a..19cd0937e740 100644
--- a/sound/soc/fsl/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
@@ -1,8 +1,4 @@
-/*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+/* SPDX-License-Identifier: GPL-2.0 */
 
 #ifndef _IMX_SSI_H
 #define _IMX_SSI_H
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index c1a4544eb16b..ccf9301889fe 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -1,10 +1,10 @@
-/*
- * Freescale MPC5200 PSC DMA
- * ALSA SoC Platform driver
- *
- * Copyright (C) 2008 Secret Lab Technologies Ltd.
- * Copyright (C) 2009 Jon Smirl, Digispeaker
- */
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// Freescale MPC5200 PSC DMA
+// ALSA SoC Platform driver
+//
+// Copyright (C) 2008 Secret Lab Technologies Ltd.
+// Copyright (C) 2009 Jon Smirl, Digispeaker
 
 #include <linux/module.h>
 #include <linux/of_device.h>
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 07ee355ee385..e5b9c04d1565 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -1,13 +1,9 @@
-/*
- * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
- *
- * Copyright (C) 2009 Jon Smirl, Digispeaker
- * Author: Jon Smirl <jonsmirl@gmail.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
+//
+// Copyright (C) 2009 Jon Smirl, Digispeaker
+// Author: Jon Smirl <jonsmirl@gmail.com>
 
 #include <linux/module.h>
 #include <linux/of_device.h>
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index d8232943ccb6..9bc01f374b39 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -1,10 +1,10 @@
-/*
- * Freescale MPC5200 PSC in I2S mode
- * ALSA SoC Digital Audio Interface (DAI) driver
- *
- * Copyright (C) 2008 Secret Lab Technologies Ltd.
- * Copyright (C) 2009 Jon Smirl, Digispeaker
- */
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// Freescale MPC5200 PSC in I2S mode
+// ALSA SoC Digital Audio Interface (DAI) driver
+//
+// Copyright (C) 2008 Secret Lab Technologies Ltd.
+// Copyright (C) 2009 Jon Smirl, Digispeaker
 
 #include <linux/module.h>
 #include <linux/of_device.h>
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a639b52c16f6..f6261a3eeb0f 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -1,14 +1,10 @@
-/**
- * Freescale MPC8610HPCD ALSA SoC Machine driver
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2007-2010 Freescale Semiconductor, Inc.
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale MPC8610HPCD ALSA SoC Machine driver
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2007-2010 Freescale Semiconductor, Inc.
 
 #include <linux/module.h>
 #include <linux/interrupt.h>
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index d7ec3d20065c..37a4520aef62 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -1,25 +1,10 @@
-/*
- * mx27vis-aic32x4.c
- *
- * Copyright 2011 Vista Silicon S.L.
- *
- * Author: Javier Martin <javier.martin@vista-silicon.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;  either version 2 of the  License, or (at your
- * option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
- * MA 02110-1301, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// mx27vis-aic32x4.c
+//
+// Copyright 2011 Vista Silicon S.L.
+//
+// Author: Javier Martin <javier.martin@vista-silicon.com>
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 41c623c55c16..80384f70878d 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -1,14 +1,10 @@
-/**
- * Freescale P1022DS ALSA SoC Machine driver
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2010 Freescale Semiconductor, Inc.
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale P1022DS ALSA SoC Machine driver
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2010 Freescale Semiconductor, Inc.
 
 #include <linux/module.h>
 #include <linux/fsl/guts.h>
diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c
index 4afbdd610bfa..1c32c2d8c6b0 100644
--- a/sound/soc/fsl/p1022_rdk.c
+++ b/sound/soc/fsl/p1022_rdk.c
@@ -1,21 +1,17 @@
-/**
- * Freescale P1022RDK ALSA SoC Machine driver
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2012 Freescale Semiconductor, Inc.
- *
- * 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.
- *
- * Note: in order for audio to work correctly, the output controls need
- * to be enabled, because they control the clock.  So for playback, for
- * example:
- *
- *      amixer sset 'Left Output Mixer PCM' on
- *      amixer sset 'Right Output Mixer PCM' on
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale P1022RDK ALSA SoC Machine driver
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2012 Freescale Semiconductor, Inc.
+//
+// Note: in order for audio to work correctly, the output controls need
+// to be enabled, because they control the clock.  So for playback, for
+// example:
+//
+//      amixer sset 'Left Output Mixer PCM' on
+//      amixer sset 'Right Output Mixer PCM' on
 
 #include <linux/module.h>
 #include <linux/fsl/guts.h>
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index e339f36cea95..a7fe4ad25c52 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -1,14 +1,10 @@
-/*
- * Phytec pcm030 driver for the PSC of the Freescale MPC52xx
- * configured as AC97 interface
- *
- * Copyright 2008 Jon Smirl, Digispeaker
- * Author: Jon Smirl <jonsmirl@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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Phytec pcm030 driver for the PSC of the Freescale MPC52xx
+// configured as AC97 interface
+//
+// Copyright 2008 Jon Smirl, Digispeaker
+// Author: Jon Smirl <jonsmirl@gmail.com>
 
 #include <linux/init.h>
 #include <linux/module.h>
diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c
index 66fb6c4614d2..fe7ba6db7c96 100644
--- a/sound/soc/fsl/phycore-ac97.c
+++ b/sound/soc/fsl/phycore-ac97.c
@@ -1,14 +1,8 @@
-/*
- * phycore-ac97.c  --  SoC audio for imx_phycore in AC97 mode
- *
- * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
- *
- *  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;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// phycore-ac97.c  --  SoC audio for imx_phycore in AC97 mode
+//
+// Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index 2f80b21b2921..aad24ccbef90 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -1,16 +1,11 @@
-/*
- *  wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
- *
- *  Copyright (c) 2010 Wolfson Microelectronics plc
- *  Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- *
- *  Based on an earlier driver for the same hardware by Liam Girdwood.
- *
- *  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;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+//  wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
+//
+//  Copyright (c) 2010 Wolfson Microelectronics plc
+//  Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+//
+//  Based on an earlier driver for the same hardware by Liam Girdwood.
 
 #include <linux/platform_device.h>
 #include <linux/clk.h>