[PATCH v4 0/3] Add NXP AUDMIX device and machine drivers

[PATCH v4 0/3] Add NXP AUDMIX device and machine drivers

[Viorel Suman]

The patchset adds NXP Audio Mixer (AUDMIX) device and machine
drivers and related DT bindings documentation.

Changes since V3:
1. Removed machine driver DT bindings documentation.
2. Trigger machine driver probe from device driver as suggested by Nicolin.

Changes since V2:
1. Moved "dais" node from machine driver DTS node to device driver DTS node
  as suggested by Rob.

Changes since V1:
1. Original patch split into distinct patches for the device driver and
  DT binding documentation.
2. Replaced AMIX with AUDMIX in both code and file names as it looks more
  RM-compliant.
3. Removed polarity control from CPU DAI driver as suggested by Nicolin.
4. Added machine driver and related DT binding documentation.

Viorel Suman (3):
  ASoC: fsl: Add Audio Mixer CPU DAI driver
  ASoC: add fsl_audmix DT binding documentation
  ASoC: fsl: Add Audio Mixer machine driver

 .../devicetree/bindings/sound/fsl,audmix.txt       |  54 ++
 sound/soc/fsl/Kconfig                              |  16 +
 sound/soc/fsl/Makefile                             |   5 +
 sound/soc/fsl/fsl_audmix.c                         | 576 +++++++++++++++++++++
 sound/soc/fsl/fsl_audmix.h                         | 102 ++++
 sound/soc/fsl/imx-audmix.c                         | 327 ++++++++++++
 6 files changed, 1080 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt
 create mode 100644 sound/soc/fsl/fsl_audmix.c
 create mode 100644 sound/soc/fsl/fsl_audmix.h
 create mode 100644 sound/soc/fsl/imx-audmix.c

-- 
2.7.4

[PATCH v4 1/3] ASoC: fsl: Add Audio Mixer CPU DAI driver

[Viorel Suman]

This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs.
The Audio Mixer is a on-chip functional module that allows mixing of
two audio streams into a single audio stream.

Audio Mixer datasheet is available here:
https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf

Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
---
 sound/soc/fsl/Kconfig      |   7 +
 sound/soc/fsl/Makefile     |   3 +
 sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_audmix.h | 102 ++++++++
 4 files changed, 688 insertions(+)
 create mode 100644 sound/soc/fsl/fsl_audmix.c
 create mode 100644 sound/soc/fsl/fsl_audmix.h

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 7b1d997..0af2e056 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
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 3c0ff31..4172d5a 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
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
new file mode 100644
index 0000000..3356cb6
--- /dev/null
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -0,0 +1,576 @@
+// 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, 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 int fsl_audmix_probe(struct platform_device *pdev)
+{
+	struct fsl_audmix *priv;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+	const char *sprop;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	/* Get the addresses */
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs,
+						 &fsl_audmix_regmap_config);
+	if (IS_ERR(priv->regmap)) {
+		dev_err(&pdev->dev, "failed to init regmap\n");
+		return PTR_ERR(priv->regmap);
+	}
+
+	priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg");
+	if (IS_ERR(priv->ipg_clk)) {
+		dev_err(&pdev->dev, "failed to get ipg clock\n");
+		return PTR_ERR(priv->ipg_clk);
+	}
+
+	platform_set_drvdata(pdev, priv);
+	pm_runtime_enable(&pdev->dev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_audmix_component,
+					      fsl_audmix_dai,
+					      ARRAY_SIZE(fsl_audmix_dai));
+	if (ret) {
+		dev_err(&pdev->dev, "failed to register ASoC DAI\n");
+		return ret;
+	}
+
+	sprop = of_get_property(pdev->dev.of_node, "model", NULL);
+	if (sprop) {
+		priv->pdev = platform_device_register_data(&pdev->dev, sprop, 0,
+							   NULL, 0);
+		if (IS_ERR(priv->pdev)) {
+			ret = PTR_ERR(priv->pdev);
+			dev_err(&pdev->dev,
+				"failed to register platform %s: %d\n", sprop,
+				 ret);
+		}
+	} else {
+		dev_err(&pdev->dev, "[model] attribute missing.\n");
+		ret = -EINVAL;
+	}
+
+	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 const struct of_device_id fsl_audmix_ids[] = {
+	{ .compatible = "fsl,imx8qm-audmix", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_audmix_ids);
+
+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 0000000..7812ffe
--- /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 */
-- 
2.7.4

Applied "ASoC: fsl: Add Audio Mixer CPU DAI driver" to the asoc tree

[Mark Brown]

The patch

   ASoC: fsl: Add Audio Mixer CPU DAI driver

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From be1df61cf06efb355c90702e46b8d46f055acb4e Mon Sep 17 00:00:00 2001
From: Viorel Suman <viorel.suman@nxp.com>
Date: Tue, 22 Jan 2019 11:14:26 +0000
Subject: [PATCH] ASoC: fsl: Add Audio Mixer CPU DAI driver

This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs.
The Audio Mixer is a on-chip functional module that allows mixing of
two audio streams into a single audio stream.

Audio Mixer datasheet is available here:
https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf

Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/Kconfig      |   7 +
 sound/soc/fsl/Makefile     |   3 +
 sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_audmix.h | 102 +++++++
 4 files changed, 688 insertions(+)
 create mode 100644 sound/soc/fsl/fsl_audmix.c
 create mode 100644 sound/soc/fsl/fsl_audmix.h

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 7b1d9970be8b..0af2e056d211 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
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 3c0ff315b971..4172d5a3e36c 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
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
new file mode 100644
index 000000000000..3356cb617713
--- /dev/null
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -0,0 +1,576 @@
+// 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, 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 int fsl_audmix_probe(struct platform_device *pdev)
+{
+	struct fsl_audmix *priv;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+	const char *sprop;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	/* Get the addresses */
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs,
+						 &fsl_audmix_regmap_config);
+	if (IS_ERR(priv->regmap)) {
+		dev_err(&pdev->dev, "failed to init regmap\n");
+		return PTR_ERR(priv->regmap);
+	}
+
+	priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg");
+	if (IS_ERR(priv->ipg_clk)) {
+		dev_err(&pdev->dev, "failed to get ipg clock\n");
+		return PTR_ERR(priv->ipg_clk);
+	}
+
+	platform_set_drvdata(pdev, priv);
+	pm_runtime_enable(&pdev->dev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_audmix_component,
+					      fsl_audmix_dai,
+					      ARRAY_SIZE(fsl_audmix_dai));
+	if (ret) {
+		dev_err(&pdev->dev, "failed to register ASoC DAI\n");
+		return ret;
+	}
+
+	sprop = of_get_property(pdev->dev.of_node, "model", NULL);
+	if (sprop) {
+		priv->pdev = platform_device_register_data(&pdev->dev, sprop, 0,
+							   NULL, 0);
+		if (IS_ERR(priv->pdev)) {
+			ret = PTR_ERR(priv->pdev);
+			dev_err(&pdev->dev,
+				"failed to register platform %s: %d\n", sprop,
+				 ret);
+		}
+	} else {
+		dev_err(&pdev->dev, "[model] attribute missing.\n");
+		ret = -EINVAL;
+	}
+
+	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 const struct of_device_id fsl_audmix_ids[] = {
+	{ .compatible = "fsl,imx8qm-audmix", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_audmix_ids);
+
+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 */
-- 
2.20.1

[PATCH v4 2/3] ASoC: add fsl_audmix DT binding documentation

[Viorel Suman]

Add the DT binding documentation for NXP Audio Mixer
CPU DAI driver.

Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
---
 .../devicetree/bindings/sound/fsl,audmix.txt       | 54 ++++++++++++++++++++++
 1 file changed, 54 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt

diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
new file mode 100644
index 0000000..45f807e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
@@ -0,0 +1,54 @@
+NXP Audio Mixer (AUDMIX).
+
+The Audio Mixer is a on-chip functional module that allows mixing of two
+audio streams into a single audio stream. Audio Mixer has two input serial
+audio interfaces. These are driven by two Synchronous Audio interface
+modules (SAI). Each input serial interface carries 8 audio channels in its
+frame in TDM manner. Mixer mixes audio samples of corresponding channels
+from two interfaces into a single sample. Before mixing, audio samples of
+two inputs can be attenuated based on configuration. The output of the
+Audio Mixer is also a serial audio interface. Like input interfaces it has
+the same TDM frame format. This output is used to drive the serial DAC TDM
+interface of audio codec and also sent to the external pins along with the
+receive path of normal audio SAI module for readback by the CPU.
+
+The output of Audio Mixer can be selected from any of the three streams
+ - serial audio input 1
+ - serial audio input 2
+ - mixed audio
+
+Mixing operation is independent of audio sample rate but the two audio
+input streams must have same audio sample rate with same number of channels
+in TDM frame to be eligible for mixing.
+
+Device driver required properties:
+=================================
+  - compatible		: Compatible list, contains "fsl,imx8qm-audmix"
+
+  - reg			: Offset and length of the register set for the device.
+
+  - clocks		: Must contain an entry for each entry in clock-names.
+
+  - clock-names		: Must include the "ipg" for register access.
+
+  - power-domains	: Must contain the phandle to AUDMIX power domain node
+
+  - dais		: Must contain a list of phandles to AUDMIX connected
+			  DAIs. The current implementation requires two phandles
+			  to SAI interfaces to be provided, the first SAI in the
+			  list being used to route the AUDMIX output.
+
+  - model		: Must contain machine driver name which will configure
+			  and instantiate the appropriate audio card.
+
+Device driver configuration example:
+======================================
+  audmix: audmix@59840000 {
+    compatible = "fsl,imx8qm-audmix";
+    reg = <0x0 0x59840000 0x0 0x10000>;
+    clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
+    clock-names = "ipg";
+    power-domains = <&pd_audmix>;
+    dais = <&sai4>, <&sai5>;
+    model = "imx-audmix";
+  };
-- 
2.7.4

Re: [PATCH v4 2/3] ASoC: add fsl_audmix DT binding documentation

[Nicolin Chen]

On Tue, Jan 22, 2019 at 11:14:27AM +0000, Viorel Suman wrote:
> Add the DT binding documentation for NXP Audio Mixer
> CPU DAI driver.
> 
> Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
> ---
>  .../devicetree/bindings/sound/fsl,audmix.txt       | 54 ++++++++++++++++++++++
>  1 file changed, 54 insertions(+)
>  create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt
> 
> diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
> new file mode 100644
> index 0000000..45f807e
> --- /dev/null
> +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
> @@ -0,0 +1,54 @@
> +NXP Audio Mixer (AUDMIX).
> +
> +The Audio Mixer is a on-chip functional module that allows mixing of two
> +audio streams into a single audio stream. Audio Mixer has two input serial
> +audio interfaces. These are driven by two Synchronous Audio interface
> +modules (SAI). Each input serial interface carries 8 audio channels in its
> +frame in TDM manner. Mixer mixes audio samples of corresponding channels
> +from two interfaces into a single sample. Before mixing, audio samples of
> +two inputs can be attenuated based on configuration. The output of the
> +Audio Mixer is also a serial audio interface. Like input interfaces it has
> +the same TDM frame format. This output is used to drive the serial DAC TDM
> +interface of audio codec and also sent to the external pins along with the
> +receive path of normal audio SAI module for readback by the CPU.
> +
> +The output of Audio Mixer can be selected from any of the three streams
> + - serial audio input 1
> + - serial audio input 2
> + - mixed audio
> +
> +Mixing operation is independent of audio sample rate but the two audio
> +input streams must have same audio sample rate with same number of channels
> +in TDM frame to be eligible for mixing.
> +
> +Device driver required properties:
> +=================================
> +  - compatible		: Compatible list, contains "fsl,imx8qm-audmix"
> +
> +  - reg			: Offset and length of the register set for the device.
> +
> +  - clocks		: Must contain an entry for each entry in clock-names.
> +
> +  - clock-names		: Must include the "ipg" for register access.
> +
> +  - power-domains	: Must contain the phandle to AUDMIX power domain node
> +
> +  - dais		: Must contain a list of phandles to AUDMIX connected
> +			  DAIs. The current implementation requires two phandles
> +			  to SAI interfaces to be provided, the first SAI in the
> +			  list being used to route the AUDMIX output.
> +
> +  - model		: Must contain machine driver name which will configure
> +			  and instantiate the appropriate audio card.

I don't feel this is necessary for DT node. The driver should be
able to figure out a proper name for the machine driver: it could
use the compatible string to get "imx8qm-audmix" as the name like
fsl_ssi; Or it could have built-in "imx-audmix" of_device_id data
corresponding to "fsl,imx8qm-audmix" compatible string -- for any
non-imx compatible in the future, use something else.

Thanks
Nic

> +
> +Device driver configuration example:
> +======================================
> +  audmix: audmix@59840000 {
> +    compatible = "fsl,imx8qm-audmix";
> +    reg = <0x0 0x59840000 0x0 0x10000>;
> +    clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
> +    clock-names = "ipg";
> +    power-domains = <&pd_audmix>;
> +    dais = <&sai4>, <&sai5>;
> +    model = "imx-audmix";
> +  };
> -- 
> 2.7.4
> 

Re: [PATCH v4 2/3] ASoC: add fsl_audmix DT binding documentation

[Rob Herring]

On Fri, Jan 25, 2019 at 11:43:16AM -0800, Nicolin Chen wrote:
> On Tue, Jan 22, 2019 at 11:14:27AM +0000, Viorel Suman wrote:
> > Add the DT binding documentation for NXP Audio Mixer
> > CPU DAI driver.
> > 
> > Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
> > ---
> >  .../devicetree/bindings/sound/fsl,audmix.txt       | 54 ++++++++++++++++++++++
> >  1 file changed, 54 insertions(+)
> >  create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt
> > 
> > diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
> > new file mode 100644
> > index 0000000..45f807e
> > --- /dev/null
> > +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
> > @@ -0,0 +1,54 @@
> > +NXP Audio Mixer (AUDMIX).
> > +
> > +The Audio Mixer is a on-chip functional module that allows mixing of two
> > +audio streams into a single audio stream. Audio Mixer has two input serial
> > +audio interfaces. These are driven by two Synchronous Audio interface
> > +modules (SAI). Each input serial interface carries 8 audio channels in its
> > +frame in TDM manner. Mixer mixes audio samples of corresponding channels
> > +from two interfaces into a single sample. Before mixing, audio samples of
> > +two inputs can be attenuated based on configuration. The output of the
> > +Audio Mixer is also a serial audio interface. Like input interfaces it has
> > +the same TDM frame format. This output is used to drive the serial DAC TDM
> > +interface of audio codec and also sent to the external pins along with the
> > +receive path of normal audio SAI module for readback by the CPU.
> > +
> > +The output of Audio Mixer can be selected from any of the three streams
> > + - serial audio input 1
> > + - serial audio input 2
> > + - mixed audio
> > +
> > +Mixing operation is independent of audio sample rate but the two audio
> > +input streams must have same audio sample rate with same number of channels
> > +in TDM frame to be eligible for mixing.
> > +
> > +Device driver required properties:
> > +=================================
> > +  - compatible		: Compatible list, contains "fsl,imx8qm-audmix"
> > +
> > +  - reg			: Offset and length of the register set for the device.
> > +
> > +  - clocks		: Must contain an entry for each entry in clock-names.
> > +
> > +  - clock-names		: Must include the "ipg" for register access.
> > +
> > +  - power-domains	: Must contain the phandle to AUDMIX power domain node
> > +
> > +  - dais		: Must contain a list of phandles to AUDMIX connected
> > +			  DAIs. The current implementation requires two phandles
> > +			  to SAI interfaces to be provided, the first SAI in the
> > +			  list being used to route the AUDMIX output.
> > +
> > +  - model		: Must contain machine driver name which will configure
> > +			  and instantiate the appropriate audio card.
> 
> I don't feel this is necessary for DT node. The driver should be
> able to figure out a proper name for the machine driver: it could
> use the compatible string to get "imx8qm-audmix" as the name like
> fsl_ssi; Or it could have built-in "imx-audmix" of_device_id data
> corresponding to "fsl,imx8qm-audmix" compatible string -- for any
> non-imx compatible in the future, use something else.

I tend to agree, but either way:

Rob Herring <robh@kernel.org>

Applied "ASoC: add fsl_audmix DT binding documentation" to the asoc tree

[Mark Brown]

The patch

   ASoC: add fsl_audmix DT binding documentation

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From d0d9071b724123ebde89bdf6b52b86c3289abe85 Mon Sep 17 00:00:00 2001
From: Viorel Suman <viorel.suman@nxp.com>
Date: Tue, 22 Jan 2019 11:14:27 +0000
Subject: [PATCH] ASoC: add fsl_audmix DT binding documentation

Add the DT binding documentation for NXP Audio Mixer
CPU DAI driver.

Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/fsl,audmix.txt  | 54 +++++++++++++++++++
 1 file changed, 54 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt

diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
new file mode 100644
index 000000000000..45f807ec21a5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
@@ -0,0 +1,54 @@
+NXP Audio Mixer (AUDMIX).
+
+The Audio Mixer is a on-chip functional module that allows mixing of two
+audio streams into a single audio stream. Audio Mixer has two input serial
+audio interfaces. These are driven by two Synchronous Audio interface
+modules (SAI). Each input serial interface carries 8 audio channels in its
+frame in TDM manner. Mixer mixes audio samples of corresponding channels
+from two interfaces into a single sample. Before mixing, audio samples of
+two inputs can be attenuated based on configuration. The output of the
+Audio Mixer is also a serial audio interface. Like input interfaces it has
+the same TDM frame format. This output is used to drive the serial DAC TDM
+interface of audio codec and also sent to the external pins along with the
+receive path of normal audio SAI module for readback by the CPU.
+
+The output of Audio Mixer can be selected from any of the three streams
+ - serial audio input 1
+ - serial audio input 2
+ - mixed audio
+
+Mixing operation is independent of audio sample rate but the two audio
+input streams must have same audio sample rate with same number of channels
+in TDM frame to be eligible for mixing.
+
+Device driver required properties:
+=================================
+  - compatible		: Compatible list, contains "fsl,imx8qm-audmix"
+
+  - reg			: Offset and length of the register set for the device.
+
+  - clocks		: Must contain an entry for each entry in clock-names.
+
+  - clock-names		: Must include the "ipg" for register access.
+
+  - power-domains	: Must contain the phandle to AUDMIX power domain node
+
+  - dais		: Must contain a list of phandles to AUDMIX connected
+			  DAIs. The current implementation requires two phandles
+			  to SAI interfaces to be provided, the first SAI in the
+			  list being used to route the AUDMIX output.
+
+  - model		: Must contain machine driver name which will configure
+			  and instantiate the appropriate audio card.
+
+Device driver configuration example:
+======================================
+  audmix: audmix@59840000 {
+    compatible = "fsl,imx8qm-audmix";
+    reg = <0x0 0x59840000 0x0 0x10000>;
+    clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
+    clock-names = "ipg";
+    power-domains = <&pd_audmix>;
+    dais = <&sai4>, <&sai5>;
+    model = "imx-audmix";
+  };
-- 
2.20.1

[PATCH v4 3/3] ASoC: fsl: Add Audio Mixer machine driver

[Viorel Suman]

This patch implements Audio Mixer machine driver for NXP iMX8 SOCs.
It connects together Audio Mixer and related SAI instances.

Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
---
 sound/soc/fsl/Kconfig      |   9 ++
 sound/soc/fsl/Makefile     |   2 +
 sound/soc/fsl/imx-audmix.c | 327 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 338 insertions(+)
 create mode 100644 sound/soc/fsl/imx-audmix.c

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 0af2e056..d87c842 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -303,6 +303,15 @@ 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 4172d5a..c0dd044 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -62,6 +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
@@ -71,3 +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/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
new file mode 100644
index 0000000..72e37ca
--- /dev/null
+++ b/sound/soc/fsl/imx-audmix.c
@@ -0,0 +1,327 @@
+// 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, "Missign 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;
+	}
+
+	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;
+		}
+
+		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;
+	}
+	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");
-- 
2.7.4