[PATCH v5 0/3] Add NXP AUDMIX device and machine drivers

[PATCH v5 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 V4:
1. Removed "model" attribute from device driver DT bindings documentation
   as suggested by Nicolin.

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       |  50 ++
 sound/soc/fsl/Kconfig                              |  16 +
 sound/soc/fsl/Makefile                             |   5 +
 sound/soc/fsl/fsl_audmix.c                         | 578 +++++++++++++++++++++
 sound/soc/fsl/fsl_audmix.h                         | 102 ++++
 sound/soc/fsl/imx-audmix.c                         | 327 ++++++++++++
 6 files changed, 1078 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 v5 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 | 578 +++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_audmix.h | 102 ++++++++
 4 files changed, 690 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..07b72a3
--- /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, 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 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

[PATCH v5 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       | 50 ++++++++++++++++++++++
 1 file changed, 50 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..840b7e0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
@@ -0,0 +1,50 @@
+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.
+
+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>;
+  };
-- 
2.7.4

[PATCH v5 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

Re: [PATCH v5 0/3] Add NXP AUDMIX device and machine drivers

[Nicolin Chen]

On Fri, Feb 15, 2019 at 02:01:32PM +0000, Viorel Suman wrote:
> The patchset adds NXP Audio Mixer (AUDMIX) device and machine
> drivers and related DT bindings documentation.

For this series,

Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>

And Rob gave his at the previous version already.

Thanks.

> Changes since V4:
> 1. Removed "model" attribute from device driver DT bindings documentation
>    as suggested by Nicolin.
> 
> 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       |  50 ++
>  sound/soc/fsl/Kconfig                              |  16 +
>  sound/soc/fsl/Makefile                             |   5 +
>  sound/soc/fsl/fsl_audmix.c                         | 578 +++++++++++++++++++++
>  sound/soc/fsl/fsl_audmix.h                         | 102 ++++
>  sound/soc/fsl/imx-audmix.c                         | 327 ++++++++++++
>  6 files changed, 1078 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
>