Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became wide and scattered updates all over the sound tree as
  diffstat shows: lots of (still ongoing) refactoring works in ASoC,
  fixes and cleanups caught by static analysis, inclusive term
  conversions as well as lots of new drivers. Below are highlights:

  ASoC core:
   - API cleanups and conversions to the unified mute_stream() call
   - Simplify I/O helper functions
   - Use helper macros to retrieve RTD from substreams

  ASoC drivers:
   - Lots of fixes and cleanups in Intel ASoC drivers
   - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
     Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
     nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
     boards, TI J721e EVM

  ALSA core:
   - Minor code refacotring for SG-buffer handling

  HD-audio:
   - Generalization of mute-LED handling with LED classdev
   - Intel silent stream support for HDMI
   - Device-specific fixes: CA0132, Loongson-3

  Others:
   - Usual USB- and HD-audio quirks for various devices
   - Fixes for echoaudio DMA position handling
   - Various documents and trivial fixes for sparse warnings
   - Conversion to adopt inclusive terms"

* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
  ALSA: pci: delete repeated words in comments
  ALSA: isa: delete repeated words in comments
  ALSA: hda/tegra: Add 100us dma stop delay
  ALSA: hda: Add dma stop delay variable
  ASoC: hda/tegra: Set buffer alignment to 128 bytes
  ALSA: seq: oss: Serialize ioctls
  ALSA: hda/hdmi: Add quirk to force connectivity
  ALSA: usb-audio: add startech usb audio dock name
  ALSA: usb-audio: Add support for Lenovo ThinkStation P620
  Revert "ALSA: hda: call runtime_allow() for all hda controllers"
  ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
  ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
  ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
  ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
  ALSA: docs: fix typo
  ALSA: doc: use correct config variable name
  ASoC: core: Two step component registration
  ASoC: core: Simplify snd_soc_component_initialize declaration
  ASoC: core: Relocate and expose snd_soc_component_initialize
  ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
  ...

20 files changed, 937 insertions, 29 deletions

diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index c5408c129f34..1e6ab87e4460 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -219,5 +219,14 @@ config SND_SOC_DM365_VOICE_CODEC_MODULE
 	  The is an internal symbol needed to ensure that the codec
 	  and MFD driver can be built as loadable modules if necessary.
 
+config SND_SOC_J721E_EVM
+	tristate "SoC Audio support for j721e EVM"
+	depends on ARCH_K3_J721E_SOC || COMPILE_TEST
+	depends on I2C
+	select SND_SOC_PCM3168A_I2C
+	select SND_SOC_DAVINCI_MCASP
+	help
+	  Say Y if you want to add support for SoC audio on j721e Common
+	  Processor Board and Infotainment expansion board.
 endmenu
 
diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile
index ea48c6679cc7..a21e5b0061de 100644
--- a/sound/soc/ti/Makefile
+++ b/sound/soc/ti/Makefile
@@ -34,6 +34,7 @@ snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
 snd-soc-ams-delta-objs := ams-delta.o
 snd-soc-omap-hdmi-objs := omap-hdmi.o
 snd-soc-osk5912-objs := osk5912.o
+snd-soc-j721e-evm-objs := j721e-evm.o
 
 obj-$(CONFIG_SND_SOC_DAVINCI_EVM) += snd-soc-davinci-evm.o
 obj-$(CONFIG_SND_SOC_NOKIA_N810) += snd-soc-n810.o
@@ -44,3 +45,4 @@ obj-$(CONFIG_SND_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
 obj-$(CONFIG_SND_SOC_OMAP_AMS_DELTA) += snd-soc-ams-delta.o
 obj-$(CONFIG_SND_SOC_OMAP_HDMI) += snd-soc-omap-hdmi.o
 obj-$(CONFIG_SND_SOC_OMAP_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_SOC_J721E_EVM) += snd-soc-j721e-evm.o
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index e17cd5e939f0..5c47de96c529 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -420,7 +420,7 @@ static struct snd_soc_ops ams_delta_ops;
  * Shares hardware with codec config pulse generation */
 static bool ams_delta_muted = 1;
 
-static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+static int ams_delta_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	int apply;
 
@@ -439,18 +439,19 @@ static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
 
 /* Our codec DAI probably doesn't have its own .ops structure */
 static const struct snd_soc_dai_ops ams_delta_dai_ops = {
-	.digital_mute = ams_delta_digital_mute,
+	.mute_stream = ams_delta_mute,
+	.no_capture_mute = 1,
 };
 
 /* Will be used if the codec ever has its own digital_mute function */
 static int ams_delta_startup(struct snd_pcm_substream *substream)
 {
-	return ams_delta_digital_mute(NULL, 0);
+	return ams_delta_digital_mute(NULL, 0, substream->stream);
 }
 
 static void ams_delta_shutdown(struct snd_pcm_substream *substream)
 {
-	ams_delta_digital_mute(NULL, 1);
+	ams_delta_digital_mute(NULL, 1, substream->stream);
 }
 
 
diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
index 2cfbeebdfb41..105e56ab9cdc 100644
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -28,7 +28,7 @@ struct snd_soc_card_drvdata_davinci {
 
 static int evm_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *soc_card = rtd->card;
 	struct snd_soc_card_drvdata_davinci *drvdata =
 		snd_soc_card_get_drvdata(soc_card);
@@ -41,7 +41,7 @@ static int evm_startup(struct snd_pcm_substream *substream)
 
 static void evm_shutdown(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *soc_card = rtd->card;
 	struct snd_soc_card_drvdata_davinci *drvdata =
 		snd_soc_card_get_drvdata(soc_card);
@@ -53,7 +53,7 @@ static void evm_shutdown(struct snd_pcm_substream *substream)
 static int evm_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct snd_soc_card *soc_card = rtd->card;
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index b93c1ee302c0..617440767c45 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1623,12 +1623,14 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 		.name		= "davinci-mcasp.0",
 		.probe		= davinci_mcasp_dai_probe,
 		.playback	= {
+			.stream_name = "IIS Playback",
 			.channels_min	= 1,
 			.channels_max	= 32 * 16,
 			.rates 		= DAVINCI_MCASP_RATES,
 			.formats	= DAVINCI_MCASP_PCM_FMTS,
 		},
 		.capture 	= {
+			.stream_name = "IIS Capture",
 			.channels_min 	= 1,
 			.channels_max	= 32 * 16,
 			.rates 		= DAVINCI_MCASP_RATES,
@@ -1642,6 +1644,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 		.name		= "davinci-mcasp.1",
 		.probe		= davinci_mcasp_dai_probe,
 		.playback 	= {
+			.stream_name = "DIT Playback",
 			.channels_min	= 1,
 			.channels_max	= 384,
 			.rates		= DAVINCI_MCASP_RATES,
diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c
index ee4d3ef821a1..f810123cc407 100644
--- a/sound/soc/ti/davinci-vcif.c
+++ b/sound/soc/ti/davinci-vcif.c
@@ -41,7 +41,7 @@ struct davinci_vcif_dev {
 
 static void davinci_vcif_start(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct davinci_vcif_dev *davinci_vcif_dev =
 			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
@@ -60,7 +60,7 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream)
 
 static void davinci_vcif_stop(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct davinci_vcif_dev *davinci_vcif_dev =
 			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
new file mode 100644
index 000000000000..cb074af47a7d
--- /dev/null
+++ b/sound/soc/ti/j721e-evm.c
@@ -0,0 +1,896 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *  Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "davinci-mcasp.h"
+
+/*
+ * Maximum number of configuration entries for prefixes:
+ * CPB: 2 (mcasp10 + codec)
+ * IVI: 3 (mcasp0 + 2x codec)
+ */
+#define J721E_CODEC_CONF_COUNT	5
+
+#define J721E_AUDIO_DOMAIN_CPB	0
+#define J721E_AUDIO_DOMAIN_IVI	1
+
+#define J721E_CLK_PARENT_48000	0
+#define J721E_CLK_PARENT_44100	1
+
+#define J721E_MAX_CLK_HSDIV	128
+#define PCM1368A_MAX_SYSCLK	36864000
+
+#define J721E_DAI_FMT		(SND_SOC_DAIFMT_RIGHT_J | \
+				 SND_SOC_DAIFMT_NB_NF |   \
+				 SND_SOC_DAIFMT_CBS_CFS)
+
+enum j721e_board_type {
+	J721E_BOARD_CPB = 1,
+	J721E_BOARD_CPB_IVI,
+};
+
+struct j721e_audio_match_data {
+	enum j721e_board_type board_type;
+	int num_links;
+	unsigned int pll_rates[2];
+};
+
+static unsigned int ratios_for_pcm3168a[] = {
+	256,
+	512,
+	768,
+};
+
+struct j721e_audio_clocks {
+	struct clk *target;
+	struct clk *parent[2];
+};
+
+struct j721e_audio_domain {
+	struct j721e_audio_clocks codec;
+	struct j721e_audio_clocks mcasp;
+	int parent_clk_id;
+
+	int active;
+	unsigned int active_link;
+	unsigned int rate;
+};
+
+struct j721e_priv {
+	struct device *dev;
+	struct snd_soc_card card;
+	struct snd_soc_dai_link *dai_links;
+	struct snd_soc_codec_conf codec_conf[J721E_CODEC_CONF_COUNT];
+	struct snd_interval rate_range;
+	const struct j721e_audio_match_data *match_data;
+	u32 pll_rates[2];
+	unsigned int hsdiv_rates[2];
+
+	struct j721e_audio_domain audio_domains[2];
+
+	struct mutex mutex;
+};
+
+static const struct snd_soc_dapm_widget j721e_cpb_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("CPB Stereo HP 1", NULL),
+	SND_SOC_DAPM_HP("CPB Stereo HP 2", NULL),
+	SND_SOC_DAPM_HP("CPB Stereo HP 3", NULL),
+	SND_SOC_DAPM_LINE("CPB Line Out", NULL),
+	SND_SOC_DAPM_MIC("CPB Stereo Mic 1", NULL),
+	SND_SOC_DAPM_MIC("CPB Stereo Mic 2", NULL),
+	SND_SOC_DAPM_LINE("CPB Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_cpb_dapm_routes[] = {
+	{"CPB Stereo HP 1", NULL, "codec-1 AOUT1L"},
+	{"CPB Stereo HP 1", NULL, "codec-1 AOUT1R"},
+	{"CPB Stereo HP 2", NULL, "codec-1 AOUT2L"},
+	{"CPB Stereo HP 2", NULL, "codec-1 AOUT2R"},
+	{"CPB Stereo HP 3", NULL, "codec-1 AOUT3L"},
+	{"CPB Stereo HP 3", NULL, "codec-1 AOUT3R"},
+	{"CPB Line Out", NULL, "codec-1 AOUT4L"},
+	{"CPB Line Out", NULL, "codec-1 AOUT4R"},
+
+	{"codec-1 AIN1L", NULL, "CPB Stereo Mic 1"},
+	{"codec-1 AIN1R", NULL, "CPB Stereo Mic 1"},
+	{"codec-1 AIN2L", NULL, "CPB Stereo Mic 2"},
+	{"codec-1 AIN2R", NULL, "CPB Stereo Mic 2"},
+	{"codec-1 AIN3L", NULL, "CPB Line In"},
+	{"codec-1 AIN3R", NULL, "CPB Line In"},
+};
+
+static const struct snd_soc_dapm_widget j721e_ivi_codec_a_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("IVI A Line Out 1", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line Out 2", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line Out 3", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line Out 4", NULL),
+	SND_SOC_DAPM_MIC("IVI A Stereo Mic 1", NULL),
+	SND_SOC_DAPM_MIC("IVI A Stereo Mic 2", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_codec_a_dapm_routes[] = {
+	{"IVI A Line Out 1", NULL, "codec-a AOUT1L"},
+	{"IVI A Line Out 1", NULL, "codec-a AOUT1R"},
+	{"IVI A Line Out 2", NULL, "codec-a AOUT2L"},
+	{"IVI A Line Out 2", NULL, "codec-a AOUT2R"},
+	{"IVI A Line Out 3", NULL, "codec-a AOUT3L"},
+	{"IVI A Line Out 3", NULL, "codec-a AOUT3R"},
+	{"IVI A Line Out 4", NULL, "codec-a AOUT4L"},
+	{"IVI A Line Out 4", NULL, "codec-a AOUT4R"},
+
+	{"codec-a AIN1L", NULL, "IVI A Stereo Mic 1"},
+	{"codec-a AIN1R", NULL, "IVI A Stereo Mic 1"},
+	{"codec-a AIN2L", NULL, "IVI A Stereo Mic 2"},
+	{"codec-a AIN2R", NULL, "IVI A Stereo Mic 2"},
+	{"codec-a AIN3L", NULL, "IVI A Line In"},
+	{"codec-a AIN3R", NULL, "IVI A Line In"},
+};
+
+static const struct snd_soc_dapm_widget j721e_ivi_codec_b_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("IVI B Line Out 1", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line Out 2", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line Out 3", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line Out 4", NULL),
+	SND_SOC_DAPM_MIC("IVI B Stereo Mic 1", NULL),
+	SND_SOC_DAPM_MIC("IVI B Stereo Mic 2", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_codec_b_dapm_routes[] = {
+	{"IVI B Line Out 1", NULL, "codec-b AOUT1L"},
+	{"IVI B Line Out 1", NULL, "codec-b AOUT1R"},
+	{"IVI B Line Out 2", NULL, "codec-b AOUT2L"},
+	{"IVI B Line Out 2", NULL, "codec-b AOUT2R"},
+	{"IVI B Line Out 3", NULL, "codec-b AOUT3L"},
+	{"IVI B Line Out 3", NULL, "codec-b AOUT3R"},
+	{"IVI B Line Out 4", NULL, "codec-b AOUT4L"},
+	{"IVI B Line Out 4", NULL, "codec-b AOUT4R"},
+
+	{"codec-b AIN1L", NULL, "IVI B Stereo Mic 1"},
+	{"codec-b AIN1R", NULL, "IVI B Stereo Mic 1"},
+	{"codec-b AIN2L", NULL, "IVI B Stereo Mic 2"},
+	{"codec-b AIN2R", NULL, "IVI B Stereo Mic 2"},
+	{"codec-b AIN3L", NULL, "IVI B Line In"},
+	{"codec-b AIN3R", NULL, "IVI B Line In"},
+};
+
+static int j721e_configure_refclk(struct j721e_priv *priv,
+				  unsigned int audio_domain, unsigned int rate)
+{
+	struct j721e_audio_domain *domain = &priv->audio_domains[audio_domain];
+	unsigned int scki;
+	int ret = -EINVAL;
+	int i, clk_id;
+
+	if (!(rate % 8000) && priv->pll_rates[J721E_CLK_PARENT_48000])
+		clk_id = J721E_CLK_PARENT_48000;
+	else if (!(rate % 11025) && priv->pll_rates[J721E_CLK_PARENT_44100])
+		clk_id = J721E_CLK_PARENT_44100;
+	else
+		return ret;
+
+	for (i = 0; i < ARRAY_SIZE(ratios_for_pcm3168a); i++) {
+		scki = ratios_for_pcm3168a[i] * rate;
+
+		if (priv->pll_rates[clk_id] / scki <= J721E_MAX_CLK_HSDIV) {
+			ret = 0;
+			break;
+		}
+	}
+
+	if (ret) {
+		dev_err(priv->dev, "No valid clock configuration for %u Hz\n",
+			rate);
+		return ret;
+	}
+
+	if (priv->hsdiv_rates[domain->parent_clk_id] != scki) {
+		dev_dbg(priv->dev,
+			"%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
+			audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI",
+			rate,
+			clk_id == J721E_CLK_PARENT_48000 ? "PLL4" : "PLL15",
+			ratios_for_pcm3168a[i], scki);
+
+		if (domain->parent_clk_id != clk_id) {
+			ret = clk_set_parent(domain->codec.target,
+					     domain->codec.parent[clk_id]);
+			if (ret)
+				return ret;
+
+			ret = clk_set_parent(domain->mcasp.target,
+					     domain->mcasp.parent[clk_id]);
+			if (ret)
+				return ret;
+
+			domain->parent_clk_id = clk_id;
+		}
+
+		ret = clk_set_rate(domain->codec.target, scki);
+		if (ret) {
+			dev_err(priv->dev, "codec set rate failed for %u Hz\n",
+				scki);
+			return ret;
+		}
+
+		ret = clk_set_rate(domain->mcasp.target, scki);
+		if (!ret) {
+			priv->hsdiv_rates[domain->parent_clk_id] = scki;
+		} else {
+			dev_err(priv->dev, "mcasp set rate failed for %u Hz\n",
+				scki);
+			return ret;
+		}
+	}
+
+	return ret;
+}
+
+static int j721e_rule_rate(struct snd_pcm_hw_params *params,
+			   struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *t = rule->private;
+
+	return snd_interval_refine(hw_param_interval(params, rule->var), t);
+}
+
+static int j721e_audio_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
+	unsigned int active_rate;
+	int ret = 0;
+	int i;
+
+	mutex_lock(&priv->mutex);
+
+	domain->active++;
+
+	if (priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].rate)
+		active_rate = priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].rate;
+	else
+		active_rate = priv->audio_domains[J721E_AUDIO_DOMAIN_IVI].rate;
+
+	if (active_rate)
+		ret = snd_pcm_hw_constraint_single(substream->runtime,
+						   SNDRV_PCM_HW_PARAM_RATE,
+						   active_rate);
+	else
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_RATE,
+					  j721e_rule_rate, &priv->rate_range,
+					  SNDRV_PCM_HW_PARAM_RATE, -1);
+
+	mutex_unlock(&priv->mutex);
+
+	if (ret)
+		return ret;
+
+	/* Reset TDM slots to 32 */
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
+	if (ret && ret != -ENOTSUPP)
+		return ret;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int j721e_audio_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_card *card = rtd->card;
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
+	unsigned int sysclk_rate;
+	int slot_width = 32;
+	int ret;
+	int i;
+
+	mutex_lock(&priv->mutex);
+
+	if (domain->rate && domain->rate != params_rate(params)) {
+		ret = -EINVAL;
+		goto out;
+	}
+
+	if (params_width(params) == 16)
+		slot_width = 16;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, slot_width);
+	if (ret && ret != -ENOTSUPP)
+		goto out;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2,
+					       slot_width);
+		if (ret && ret != -ENOTSUPP)
+			goto out;
+	}
+
+	ret = j721e_configure_refclk(priv, domain_id, params_rate(params));
+	if (ret)
+		goto out;
+
+	sysclk_rate = priv->hsdiv_rates[domain->parent_clk_id];
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk_rate,
+					     SND_SOC_CLOCK_IN);
+		if (ret && ret != -ENOTSUPP) {
+			dev_err(priv->dev,
+				"codec set_sysclk failed for %u Hz\n",
+				sysclk_rate);
+			goto out;
+		}
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, MCASP_CLK_HCLK_AUXCLK,
+				     sysclk_rate, SND_SOC_CLOCK_IN);
+
+	if (ret && ret != -ENOTSUPP) {
+		dev_err(priv->dev, "mcasp set_sysclk failed for %u Hz\n",
+			sysclk_rate);
+	} else {
+		domain->rate = params_rate(params);
+		ret = 0;
+	}
+
+out:
+	mutex_unlock(&priv->mutex);
+	return ret;
+}
+
+static void j721e_audio_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+
+	mutex_lock(&priv->mutex);
+
+	domain->active--;
+	if (!domain->active) {
+		domain->rate = 0;
+		domain->active_link = 0;
+	}
+
+	mutex_unlock(&priv->mutex);
+}
+
+static const struct snd_soc_ops j721e_audio_ops = {
+	.startup = j721e_audio_startup,
+	.hw_params = j721e_audio_hw_params,
+	.shutdown = j721e_audio_shutdown,
+};
+
+static int j721e_audio_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
+	unsigned int sysclk_rate;
+	int i, ret;
+
+	/* Set up initial clock configuration */
+	ret = j721e_configure_refclk(priv, domain_id, 48000);
+	if (ret)
+		return ret;
+
+	sysclk_rate = priv->hsdiv_rates[domain->parent_clk_id];
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk_rate,
+					     SND_SOC_CLOCK_IN);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, MCASP_CLK_HCLK_AUXCLK,
+				     sysclk_rate, SND_SOC_CLOCK_IN);
+	if (ret && ret != -ENOTSUPP)
+		return ret;
+
+	/* Set initial tdm slots */
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
+	if (ret && ret != -ENOTSUPP)
+		return ret;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int j721e_audio_init_ivi(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
+
+	snd_soc_dapm_new_controls(dapm, j721e_ivi_codec_a_dapm_widgets,
+				  ARRAY_SIZE(j721e_ivi_codec_a_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, j721e_codec_a_dapm_routes,
+				ARRAY_SIZE(j721e_codec_a_dapm_routes));
+	snd_soc_dapm_new_controls(dapm, j721e_ivi_codec_b_dapm_widgets,
+				  ARRAY_SIZE(j721e_ivi_codec_b_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, j721e_codec_b_dapm_routes,
+				ARRAY_SIZE(j721e_codec_b_dapm_routes));
+
+	return j721e_audio_init(rtd);
+}
+
+static int j721e_get_clocks(struct device *dev,
+			    struct j721e_audio_clocks *clocks, char *prefix)
+{
+	struct clk *parent;
+	char *clk_name;
+	int ret;
+
+	clocks->target = devm_clk_get(dev, prefix);
+	if (IS_ERR(clocks->target)) {
+		ret = PTR_ERR(clocks->target);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to acquire %s: %d\n",
+				prefix, ret);
+		return ret;
+	}
+
+	clk_name = kasprintf(GFP_KERNEL, "%s-48000", prefix);
+	if (clk_name) {
+		parent = devm_clk_get(dev, clk_name);
+		kfree(clk_name);
+		if (IS_ERR(parent)) {
+			ret = PTR_ERR(parent);
+			if (ret == -EPROBE_DEFER)
+				return ret;
+
+			dev_dbg(dev, "no 48KHz parent for %s: %d\n", prefix, ret);
+			parent = NULL;
+		}
+		clocks->parent[J721E_CLK_PARENT_48000] = parent;
+	} else {
+		return -ENOMEM;
+	}
+
+	clk_name = kasprintf(GFP_KERNEL, "%s-44100", prefix);
+	if (clk_name) {
+		parent = devm_clk_get(dev, clk_name);
+		kfree(clk_name);
+		if (IS_ERR(parent)) {
+			ret = PTR_ERR(parent);
+			if (ret == -EPROBE_DEFER)
+				return ret;
+
+			dev_dbg(dev, "no 44.1KHz parent for %s: %d\n", prefix, ret);
+			parent = NULL;
+		}
+		clocks->parent[J721E_CLK_PARENT_44100] = parent;
+	} else {
+		return -ENOMEM;
+	}
+
+	if (!clocks->parent[J721E_CLK_PARENT_44100] &&
+	    !clocks->parent[J721E_CLK_PARENT_48000]) {
+		dev_err(dev, "At least one parent clock is needed for %s\n",
+			prefix);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct j721e_audio_match_data j721e_cpb_data = {
+	.board_type = J721E_BOARD_CPB,
+	.num_links = 2, /* CPB pcm3168a */
+	.pll_rates = {
+		[J721E_CLK_PARENT_44100] = 1083801600, /* PLL15 */
+		[J721E_CLK_PARENT_48000] = 1179648000, /* PLL4 */
+	},
+};
+
+static const struct j721e_audio_match_data j721e_cpb_ivi_data = {
+	.board_type = J721E_BOARD_CPB_IVI,
+	.num_links = 4, /* CPB pcm3168a + 2x pcm3168a on IVI */
+	.pll_rates = {
+		[J721E_CLK_PARENT_44100] = 1083801600, /* PLL15 */
+		[J721E_CLK_PARENT_48000] = 1179648000, /* PLL4 */
+	},
+};
+
+static const struct of_device_id j721e_audio_of_match[] = {
+	{
+		.compatible = "ti,j721e-cpb-audio",
+		.data = &j721e_cpb_data,
+	}, {
+		.compatible = "ti,j721e-cpb-ivi-audio",
+		.data = &j721e_cpb_ivi_data,
+	},
+	{ },
+};
+MODULE_DEVICE_TABLE(of, j721e_audio_of_match);
+
+static int j721e_calculate_rate_range(struct j721e_priv *priv)
+{
+	const struct j721e_audio_match_data *match_data = priv->match_data;
+	struct j721e_audio_clocks *domain_clocks;
+	unsigned int min_rate, max_rate, pll_rate;
+	struct clk *pll;
+
+	domain_clocks = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].mcasp;
+
+	pll = clk_get_parent(domain_clocks->parent[J721E_CLK_PARENT_44100]);
+	if (IS_ERR_OR_NULL(pll)) {
+		priv->pll_rates[J721E_CLK_PARENT_44100] =
+				match_data->pll_rates[J721E_CLK_PARENT_44100];
+	} else {
+		priv->pll_rates[J721E_CLK_PARENT_44100] = clk_get_rate(pll);
+		clk_put(pll);
+	}
+
+	pll = clk_get_parent(domain_clocks->parent[J721E_CLK_PARENT_48000]);
+	if (IS_ERR_OR_NULL(pll)) {
+		priv->pll_rates[J721E_CLK_PARENT_48000] =
+				match_data->pll_rates[J721E_CLK_PARENT_48000];
+	} else {
+		priv->pll_rates[J721E_CLK_PARENT_48000] = clk_get_rate(pll);
+		clk_put(pll);
+	}
+
+	if (!priv->pll_rates[J721E_CLK_PARENT_44100] &&
+	    !priv->pll_rates[J721E_CLK_PARENT_48000]) {
+		dev_err(priv->dev, "At least one PLL is needed\n");
+		return -EINVAL;
+	}
+
+	if (priv->pll_rates[J721E_CLK_PARENT_44100])
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_44100];
+	else
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_48000];
+
+	min_rate = pll_rate / J721E_MAX_CLK_HSDIV;
+	min_rate /= ratios_for_pcm3168a[ARRAY_SIZE(ratios_for_pcm3168a) - 1];
+
+	if (priv->pll_rates[J721E_CLK_PARENT_48000])
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_48000];
+	else
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_44100];
+
+	if (pll_rate > PCM1368A_MAX_SYSCLK)
+		pll_rate = PCM1368A_MAX_SYSCLK;
+
+	max_rate = pll_rate / ratios_for_pcm3168a[0];
+
+	snd_interval_any(&priv->rate_range);
+	priv->rate_range.min = min_rate;
+	priv->rate_range.max = max_rate;
+
+	return 0;
+}
+
+static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx,
+			       int *conf_idx)
+{
+	struct device_node *node = priv->dev->of_node;
+	struct snd_soc_dai_link_component *compnent;
+	struct device_node *dai_node, *codec_node;
+	struct j721e_audio_domain *domain;
+	int comp_count, comp_idx;
+	int ret;
+
+	dai_node = of_parse_phandle(node, "ti,cpb-mcasp", 0);
+	if (!dai_node) {
+		dev_err(priv->dev, "CPB McASP node is not provided\n");
+		return -EINVAL;
+	}
+
+	codec_node = of_parse_phandle(node, "ti,cpb-codec", 0);
+	if (!codec_node) {
+		dev_err(priv->dev, "CPB codec node is not provided\n");
+		return -EINVAL;
+	}
+
+	domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB];
+	ret = j721e_get_clocks(priv->dev, &domain->codec, "cpb-codec-scki");
+	if (ret)
+		return ret;
+
+	ret = j721e_get_clocks(priv->dev, &domain->mcasp, "cpb-mcasp-auxclk");
+	if (ret)
+		return ret;
+
+	/*
+	 * Common Processor Board, two links
+	 * Link 1: McASP10 -> pcm3168a_1 DAC
+	 * Link 2: McASP10 <- pcm3168a_1 ADC
+	 */
+	comp_count = 6;
+	compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
+				GFP_KERNEL);
+	if (!compnent)
+		return -ENOMEM;
+
+	comp_idx = 0;
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_codecs = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+
+	priv->dai_links[*link_idx].name = "CPB PCM3168A Playback";
+	priv->dai_links[*link_idx].stream_name = "CPB PCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs->of_node = codec_node;
+	priv->dai_links[*link_idx].codecs->dai_name = "pcm3168a-dac";
+	priv->dai_links[*link_idx].playback_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_CPB;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_codecs = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+
+	priv->dai_links[*link_idx].name = "CPB PCM3168A Capture";
+	priv->dai_links[*link_idx].stream_name = "CPB PCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs->of_node = codec_node;
+	priv->dai_links[*link_idx].codecs->dai_name = "pcm3168a-adc";
+	priv->dai_links[*link_idx].capture_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_CPB;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = codec_node;
+	priv->codec_conf[*conf_idx].name_prefix = "codec-1";
+	(*conf_idx)++;
+	priv->codec_conf[*conf_idx].dlc.of_node = dai_node;
+	priv->codec_conf[*conf_idx].name_prefix = "McASP10";
+	(*conf_idx)++;
+
+	return 0;
+}
+
+static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx,
+			       int *conf_idx)
+{
+	struct device_node *node = priv->dev->of_node;
+	struct snd_soc_dai_link_component *compnent;
+	struct device_node *dai_node, *codeca_node, *codecb_node;
+	struct j721e_audio_domain *domain;
+	int comp_count, comp_idx;
+	int ret;
+
+	if (priv->match_data->board_type != J721E_BOARD_CPB_IVI)
+		return 0;
+
+	dai_node = of_parse_phandle(node, "ti,ivi-mcasp", 0);
+	if (!dai_node) {
+		dev_err(priv->dev, "IVI McASP node is not provided\n");
+		return -EINVAL;
+	}
+
+	codeca_node = of_parse_phandle(node, "ti,ivi-codec-a", 0);
+	if (!codeca_node) {
+		dev_err(priv->dev, "IVI codec-a node is not provided\n");
+		return -EINVAL;
+	}
+
+	codecb_node = of_parse_phandle(node, "ti,ivi-codec-b", 0);
+	if (!codecb_node) {
+		dev_warn(priv->dev, "IVI codec-b node is not provided\n");
+		return 0;
+	}
+
+	domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_IVI];
+	ret = j721e_get_clocks(priv->dev, &domain->codec, "ivi-codec-scki");
+	if (ret)
+		return ret;
+
+	ret = j721e_get_clocks(priv->dev, &domain->mcasp, "ivi-mcasp-auxclk");
+	if (ret)
+		return ret;
+
+	/*
+	 * IVI extension, two links
+	 * Link 1: McASP0 -> pcm3168a_a DAC
+	 *		  \> pcm3168a_b DAC
+	 * Link 2: McASP0 <- pcm3168a_a ADC
+	 *		   \ pcm3168a_b ADC
+	 */
+	comp_count = 8;
+	compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
+				GFP_KERNEL);
+	if (!compnent)
+		return -ENOMEM;
+
+	comp_idx = 0;
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx];
+	priv->dai_links[*link_idx].num_codecs = 2;
+	comp_idx += 2;
+
+	priv->dai_links[*link_idx].name = "IVI 2xPCM3168A Playback";
+	priv->dai_links[*link_idx].stream_name = "IVI 2xPCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs[0].of_node = codeca_node;
+	priv->dai_links[*link_idx].codecs[0].dai_name = "pcm3168a-dac";
+	priv->dai_links[*link_idx].codecs[1].of_node = codecb_node;
+	priv->dai_links[*link_idx].codecs[1].dai_name = "pcm3168a-dac";
+	priv->dai_links[*link_idx].playback_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_IVI;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init_ivi;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx];
+	priv->dai_links[*link_idx].num_codecs = 2;
+
+	priv->dai_links[*link_idx].name = "IVI 2xPCM3168A Capture";
+	priv->dai_links[*link_idx].stream_name = "IVI 2xPCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs[0].of_node = codeca_node;
+	priv->dai_links[*link_idx].codecs[0].dai_name = "pcm3168a-adc";
+	priv->dai_links[*link_idx].codecs[1].of_node = codecb_node;
+	priv->dai_links[*link_idx].codecs[1].dai_name = "pcm3168a-adc";
+	priv->dai_links[*link_idx].capture_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_IVI;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = codeca_node;
+	priv->codec_conf[*conf_idx].name_prefix = "codec-a";
+	(*conf_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = codecb_node;
+	priv->codec_conf[*conf_idx].name_prefix = "codec-b";
+	(*conf_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = dai_node;
+	priv->codec_conf[*conf_idx].name_prefix = "McASP0";
+	(*conf_idx)++;
+
+	return 0;
+}
+
+static int j721e_soc_probe(struct platform_device *pdev)
+{
+	struct device_node *node = pdev->dev.of_node;
+	struct snd_soc_card *card;
+	const struct of_device_id *match;
+	struct j721e_priv *priv;
+	int link_cnt, conf_cnt, ret;
+
+	if (!node) {
+		dev_err(&pdev->dev, "of node is missing.\n");
+		return -ENODEV;
+	}
+
+	match = of_match_node(j721e_audio_of_match, node);
+	if (!match) {
+		dev_err(&pdev->dev, "No compatible match found\n");
+		return -ENODEV;
+	}
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->match_data = match->data;
+
+	priv->dai_links = devm_kcalloc(&pdev->dev, priv->match_data->num_links,
+				       sizeof(*priv->dai_links), GFP_KERNEL);
+	if (!priv->dai_links)
+		return -ENOMEM;
+
+	priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].parent_clk_id = -1;
+	priv->audio_domains[J721E_AUDIO_DOMAIN_IVI].parent_clk_id = -1;
+	priv->dev = &pdev->dev;
+	card = &priv->card;
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dapm_widgets = j721e_cpb_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(j721e_cpb_dapm_widgets);
+	card->dapm_routes = j721e_cpb_dapm_routes;
+	card->num_dapm_routes = ARRAY_SIZE(j721e_cpb_dapm_routes);
+	card->fully_routed = 1;
+
+	if (snd_soc_of_parse_card_name(card, "model")) {
+		dev_err(&pdev->dev, "Card name is not provided\n");
+		return -ENODEV;
+	}
+
+	link_cnt = 0;
+	conf_cnt = 0;
+	ret = j721e_soc_probe_cpb(priv, &link_cnt, &conf_cnt);
+	if (ret)
+		return ret;
+
+	ret = j721e_soc_probe_ivi(priv, &link_cnt, &conf_cnt);
+	if (ret)
+		return ret;
+
+	card->dai_link = priv->dai_links;
+	card->num_links = link_cnt;
+
+	card->codec_conf = priv->codec_conf;
+	card->num_configs = conf_cnt;
+
+	ret = j721e_calculate_rate_range(priv);
+	if (ret)
+		return ret;
+
+	snd_soc_card_set_drvdata(card, priv);
+
+	mutex_init(&priv->mutex);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret)
+		dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static struct platform_driver j721e_soc_driver = {
+	.driver = {
+		.name = "j721e-audio",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = of_match_ptr(j721e_audio_of_match),
+	},
+	.probe = j721e_soc_probe,
+};
+
+module_platform_driver(j721e_soc_driver);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("ASoC machine driver for j721e Common Processor Board");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index a1672b479cb7..2802a33b9c5f 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -84,7 +84,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
 static int n810_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 
 	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
 
@@ -100,7 +100,7 @@ static void n810_shutdown(struct snd_pcm_substream *substream)
 static int n810_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int err;
 
diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c
index 61e45fea5dd8..16ea039ff865 100644
--- a/sound/soc/ti/omap-abe-twl6040.c
+++ b/sound/soc/ti/omap-abe-twl6040.c
@@ -45,7 +45,7 @@ static struct platform_device *dmic_codec_dev;
 static int omap_abe_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
@@ -77,7 +77,7 @@ static const struct snd_soc_ops omap_abe_ops = {
 static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret = 0;
 
diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c
index def2a0ce8886..3328c02f93c7 100644
--- a/sound/soc/ti/omap-hdmi.c
+++ b/sound/soc/ti/omap-hdmi.c
@@ -2,7 +2,7 @@
 /*
  * omap-hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
  *
- * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
  *
  * Author: Jyri Sarha <jsarha@ti.com>
  */
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 5a32b54bbf3b..0bc7d26c660a 100644
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -142,11 +142,8 @@ static void omap_mcbsp_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir)
 
 static void omap_mcbsp_st_chgain(struct omap_mcbsp *mcbsp)
 {
-	u16 w;
 	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
 
-	w = MCBSP_ST_READ(mcbsp, SSELCR);
-
 	MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) |
 		       ST_CH1GAIN(st_data->ch1gain));
 }
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 32e3ccdbb7a2..6025b30bbe77 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -731,7 +731,7 @@ err_st:
 static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream,
 		unsigned int packet_size)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 	int words;
@@ -896,7 +896,7 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay(
 			struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 	u16 fifo_use;
diff --git a/sound/soc/ti/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c
index 92dbe2c67290..1da05a6cdc9f 100644
--- a/sound/soc/ti/omap-twl4030.c
+++ b/sound/soc/ti/omap-twl4030.c
@@ -2,7 +2,7 @@
 /*
  * omap-twl4030.c  --  SoC audio for TI SoC based boards with twl4030 codec
  *
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2012 Texas Instruments Incorporated - https://www.ti.com
  * All rights reserved.
  *
  * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
@@ -38,7 +38,7 @@ struct omap_twl4030 {
 static int omap_twl4030_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	unsigned int fmt;
 
 	switch (params_channels(params)) {
diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c
index b04146311b31..a287e9747c2a 100644
--- a/sound/soc/ti/omap3pandora.c
+++ b/sound/soc/ti/omap3pandora.c
@@ -31,7 +31,7 @@ static struct regulator *omap3pandora_dac_reg;
 static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c
index e01485cc51a1..40e29dda7e7a 100644
--- a/sound/soc/ti/osk5912.c
+++ b/sound/soc/ti/osk5912.c
@@ -38,7 +38,7 @@ static void osk_shutdown(struct snd_pcm_substream *substream)
 static int osk_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int err;
 
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 2a714a004163..2176a95201bf 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -90,7 +90,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
 static int rx51_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 
 	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
@@ -102,7 +102,7 @@ static int rx51_startup(struct snd_pcm_substream *substream)
 static int rx51_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c
index 2b0bc234e1b6..9e7691103f05 100644
--- a/sound/soc/ti/sdma-pcm.c
+++ b/sound/soc/ti/sdma-pcm.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- *  Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ *  Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
  *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  */
 
diff --git a/sound/soc/ti/sdma-pcm.h b/sound/soc/ti/sdma-pcm.h
index cb0627c8dd34..c19efb4c043d 100644
--- a/sound/soc/ti/sdma-pcm.h
+++ b/sound/soc/ti/sdma-pcm.h
@@ -1,6 +1,6 @@
 /* SPDX-License-Identifier: GPL-2.0 */
 /*
- *  Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ *  Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
  *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  */
 
diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c
index 39830caaaf7c..2ff0f518aba5 100644
--- a/sound/soc/ti/udma-pcm.c
+++ b/sound/soc/ti/udma-pcm.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- *  Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ *  Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com
  *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  */
 
diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h
index 54111e7312c1..9ed588fd79b9 100644
--- a/sound/soc/ti/udma-pcm.h
+++ b/sound/soc/ti/udma-pcm.h
@@ -1,6 +1,6 @@
 /* SPDX-License-Identifier: GPL-2.0 */
 /*
- *  Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ *  Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
  */
 
 #ifndef __UDMA_PCM_H__