Merge branch 'asoc-5.6' into asoc-next

25 files changed, 972 insertions, 246 deletions

diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index ef20316e83d1..9ca2567d0059 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -13,6 +13,19 @@ menuconfig SND_SOC_INTEL_MACH
 
 if SND_SOC_INTEL_MACH
 
+config SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES
+	bool "Use more user friendly long card names"
+	help
+	  Some drivers report the I/O configuration to userspace through the
+	  soundcard's long card name in the control user space AP. An unfortunate
+	  side effect is that this long name may also be used by the GUI,
+	  confusing users with information they don't need.
+	  This option prevents the long name from being modified, and the I/O
+	  configuration will be provided through a different component interface.
+	  Select Y if userspace like UCM (Use Case Manager) uses the component
+	  interface.
+	  If unsure select N.
+
 if SND_SOC_INTEL_HASWELL
 
 config SND_SOC_INTEL_HASWELL_MACH
@@ -31,12 +44,27 @@ endif ## SND_SOC_INTEL_HASWELL
 
 if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL
 
+config SND_SOC_INTEL_BDW_RT5650_MACH
+	tristate "Broadwell with RT5650 codec"
+	depends on I2C
+	depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST
+	depends on X86_INTEL_LPSS || COMPILE_TEST
+	select SND_SOC_RT5645
+	help
+	  This adds the ASoC machine driver for Intel Broadwell platforms with
+	  the RT5650 codec.
+	  Say Y if you have such a device.
+	  If unsure select "N".
+
 config SND_SOC_INTEL_BDW_RT5677_MACH
 	tristate "Broadwell with RT5677 codec"
 	depends on I2C
 	depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST
 	depends on GPIOLIB || COMPILE_TEST
 	depends on X86_INTEL_LPSS || COMPILE_TEST
+	depends on SPI_MASTER
+	select SPI_PXA2XX
+	select SND_SOC_RT5677_SPI
 	select SND_SOC_RT5677
 	help
 	  This adds support for Intel Broadwell platform based boards with
@@ -261,6 +289,7 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
 	select SND_SOC_DA7219
 	select SND_SOC_MAX98357A
 	select SND_SOC_DMIC
+	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	select SND_SOC_HDAC_HDMI
 
 config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
@@ -387,6 +416,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
 	select SND_SOC_RT5682
 	select SND_SOC_MAX98357A
 	select SND_SOC_DMIC
+	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	select SND_SOC_HDAC_HDMI
 	help
 	   This adds support for ASoC machine driver for Geminilake platforms
@@ -400,6 +430,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC
 
 config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
 	tristate "SKL/KBL/BXT/APL with HDA Codecs"
+	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	select SND_SOC_HDAC_HDMI
 	select SND_SOC_DMIC
 	# SND_SOC_HDAC_HDA is already selected
@@ -419,6 +450,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
 		   (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
 	select SND_SOC_RT5682
 	select SND_SOC_DMIC
+	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	select SND_SOC_HDAC_HDMI
 	help
 	   This adds support for ASoC machine driver for SOF platforms
@@ -448,6 +480,7 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
 	select SND_SOC_RT5682
 	select SND_SOC_DMIC
 	select SND_SOC_HDAC_HDMI
+	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	help
 	  This adds support for ASoC machine driver for SOF platform with
 	  RT1011 + RT5682 I2S codec.
@@ -456,4 +489,22 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
 
 endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK
 
+if SND_SOC_SOF_JASPERLAKE
+
+config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH
+	tristate "SOF with DA7219 and MAX98373 in I2S Mode"
+	depends on I2C && ACPI
+	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	select SND_SOC_DA7219
+	select SND_SOC_MAX98373
+	select SND_SOC_DMIC
+	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
+	help
+	  This adds support for ASoC machine driver for SOF platforms
+	  with DA7219 + MAX98373 I2S audio codec.
+	  Say Y if you have such a device.
+	  If unsure select "N".
+
+endif ## SND_SOC_SOF_JASPERLAKE
+
 endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index ba1aa89db09d..b74ddd49bd39 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -2,6 +2,7 @@
 snd-soc-sst-haswell-objs := haswell.o
 snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
 snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
+snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o
 snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o
 snd-soc-sst-broadwell-objs := broadwell.o
 snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o
@@ -28,6 +29,7 @@ snd-soc-skl_rt286-objs := skl_rt286.o
 snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o hda_dsp_common.o
 snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
 snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
+snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
@@ -37,6 +39,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da721
 obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
 obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
+obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o
@@ -58,3 +61,5 @@ obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
+obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH) += snd-soc-sof_da7219_max98373.o
+
diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c
new file mode 100644
index 000000000000..1a302436d450
--- /dev/null
+++ b/sound/soc/intel/boards/bdw-rt5650.c
@@ -0,0 +1,327 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * ASoC machine driver for Intel Broadwell platforms with RT5650 codec
+ *
+ * Copyright 2019, The Chromium OS Authors.  All rights reserved.
+ */
+
+#include <linux/delay.h>
+#include <linux/gpio/consumer.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+
+#include "../common/sst-dsp.h"
+#include "../haswell/sst-haswell-ipc.h"
+
+#include "../../codecs/rt5645.h"
+
+struct bdw_rt5650_priv {
+	struct gpio_desc *gpio_hp_en;
+	struct snd_soc_component *component;
+};
+
+static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("DMIC Pair1", NULL),
+	SND_SOC_DAPM_MIC("DMIC Pair2", NULL),
+};
+
+static const struct snd_soc_dapm_route bdw_rt5650_map[] = {
+	/* Speakers */
+	{"Speaker", NULL, "SPOL"},
+	{"Speaker", NULL, "SPOR"},
+
+	/* Headset jack connectors */
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+
+	/* Digital MICs
+	 * DMIC Pair1 are the two DMICs connected on the DMICN1 connector.
+	 * DMIC Pair2 are the two DMICs connected on the DMICN2 connector.
+	 * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2
+	 * are on the right side.
+	 */
+	{"DMIC L1", NULL, "DMIC Pair1"},
+	{"DMIC R1", NULL, "DMIC Pair1"},
+	{"DMIC L2", NULL, "DMIC Pair2"},
+	{"DMIC R2", NULL, "DMIC Pair2"},
+
+	/* CODEC BE connections */
+	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static const struct snd_kcontrol_new bdw_rt5650_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("DMIC Pair1"),
+	SOC_DAPM_PIN_SWITCH("DMIC Pair2"),
+};
+
+
+static struct snd_soc_jack headphone_jack;
+static struct snd_soc_jack mic_jack;
+
+static struct snd_soc_jack_pin headphone_jack_pin = {
+	.pin	= "Headphone",
+	.mask	= SND_JACK_HEADPHONE,
+};
+
+static struct snd_soc_jack_pin mic_jack_pin = {
+	.pin	= "Headset Mic",
+	.mask	= SND_JACK_MICROPHONE,
+};
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The ADSP will covert the FE rate to 48k, max 4-channels */
+	rate->min = rate->max = 48000;
+	channels->min = 2;
+	channels->max = 4;
+
+	/* set SSP0 to 24 bit */
+	snd_mask_set_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT),
+			    SNDRV_PCM_FORMAT_S24_LE);
+
+	return 0;
+}
+
+static int bdw_rt5650_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_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	/* Workaround: set codec PLL to 19.2MHz that PLL source is
+	 * from MCLK(24MHz) to conform 2.4MHz DMIC clock.
+	 */
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
+		24000000, 19200000);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	/* The actual MCLK freq is 24MHz. The codec is told that MCLK is
+	 * 24.576MHz to satisfy the requirement of rl6231_get_clk_info.
+	 * ASRC is enabled on AD and DA filters to ensure good audio quality.
+	 */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops bdw_rt5650_ops = {
+	.hw_params = bdw_rt5650_hw_params,
+};
+
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_component *component =
+		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+	struct sst_pdata *pdata = dev_get_platdata(component->dev);
+	struct sst_hsw *broadwell = pdata->dsp;
+	int ret;
+
+	/* Set ADSP SSP port settings
+	 * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz
+	 */
+	ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+		SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+		SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4);
+	if (ret < 0) {
+		dev_err(rtd->dev, "error: failed to set device config\n");
+		return ret;
+	}
+
+	return 0;
+}
+#endif
+
+static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct bdw_rt5650_priv *bdw_rt5650 =
+		snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	/* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1.
+	 * The ASRC clock source is clk_i2s1_asrc.
+	 */
+	rt5645_sel_asrc_clk_src(component,
+				RT5645_DA_STEREO_FILTER |
+				RT5645_DA_MONO_L_FILTER |
+				RT5645_DA_MONO_R_FILTER |
+				RT5645_AD_STEREO_FILTER |
+				RT5645_AD_MONO_L_FILTER |
+				RT5645_AD_MONO_R_FILTER,
+				RT5645_CLK_SEL_I2S1_ASRC);
+
+	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
+		return ret;
+	}
+
+	/* Create and initialize headphone jack */
+	if (snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+			SND_JACK_HEADPHONE, &headphone_jack,
+			&headphone_jack_pin, 1)) {
+		dev_err(component->dev, "Can't create headphone jack\n");
+	}
+
+	/* Create and initialize mic jack */
+	if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
+			&mic_jack, &mic_jack_pin, 1)) {
+		dev_err(component->dev, "Can't create mic jack\n");
+	}
+
+	rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL);
+
+	bdw_rt5650->component = component;
+
+	return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEF(dummy,
+	DAILINK_COMP_ARRAY(COMP_DUMMY()));
+
+SND_SOC_DAILINK_DEF(fe,
+	DAILINK_COMP_ARRAY(COMP_CPU("System Pin")));
+
+SND_SOC_DAILINK_DEF(platform,
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio")));
+
+SND_SOC_DAILINK_DEF(be,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1")));
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+SND_SOC_DAILINK_DEF(ssp0_port,
+	    DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
+#else
+SND_SOC_DAILINK_DEF(ssp0_port,
+	    DAILINK_COMP_ARRAY(COMP_DUMMY()));
+#endif
+
+static struct snd_soc_dai_link bdw_rt5650_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "System PCM",
+		.stream_name = "System Playback",
+		.dynamic = 1,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+		.init = bdw_rt5650_rtd_init,
+#endif
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST,
+			SND_SOC_DPCM_TRIGGER_POST
+		},
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		SND_SOC_DAILINK_REG(fe, dummy, platform),
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "Codec",
+		.id = 0,
+		.no_pcm = 1,
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = broadwell_ssp0_fixup,
+		.ops = &bdw_rt5650_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.init = bdw_rt5650_init,
+		SND_SOC_DAILINK_REG(ssp0_port, be, platform),
+	},
+};
+
+/* ASoC machine driver for Broadwell DSP + RT5650 */
+static struct snd_soc_card bdw_rt5650_card = {
+	.name = "bdw-rt5650",
+	.owner = THIS_MODULE,
+	.dai_link = bdw_rt5650_dais,
+	.num_links = ARRAY_SIZE(bdw_rt5650_dais),
+	.dapm_widgets = bdw_rt5650_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets),
+	.dapm_routes = bdw_rt5650_map,
+	.num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map),
+	.controls = bdw_rt5650_controls,
+	.num_controls = ARRAY_SIZE(bdw_rt5650_controls),
+	.fully_routed = true,
+};
+
+static int bdw_rt5650_probe(struct platform_device *pdev)
+{
+	struct bdw_rt5650_priv *bdw_rt5650;
+	struct snd_soc_acpi_mach *mach;
+	int ret;
+
+	bdw_rt5650_card.dev = &pdev->dev;
+
+	/* Allocate driver private struct */
+	bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv),
+		GFP_KERNEL);
+	if (!bdw_rt5650)
+		return -ENOMEM;
+
+	/* override plaform name, if required */
+	mach = (&pdev->dev)->platform_data;
+	ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card,
+						    mach->mach_params.platform);
+
+	if (ret)
+		return ret;
+
+	snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650);
+
+	return devm_snd_soc_register_card(&pdev->dev, &bdw_rt5650_card);
+}
+
+static struct platform_driver bdw_rt5650_audio = {
+	.probe = bdw_rt5650_probe,
+	.driver = {
+		.name = "bdw-rt5650",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+
+module_platform_driver(bdw_rt5650_audio)
+
+/* Module information */
+MODULE_AUTHOR("Ben Zhang <benzh@chromium.org>");
+MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bdw-rt5650");
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 2af8e5a62da8..bb643c99069d 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -295,6 +295,14 @@ SND_SOC_DAILINK_DEF(platform,
 SND_SOC_DAILINK_DEF(be,
 	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-RT5677CE:00", "rt5677-aif1")));
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+SND_SOC_DAILINK_DEF(ssp0_port,
+	    DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
+#else
+SND_SOC_DAILINK_DEF(ssp0_port,
+	    DAILINK_COMP_ARRAY(COMP_DUMMY()));
+#endif
+
 /* Wake on voice interface */
 SND_SOC_DAILINK_DEFS(dsp,
 	DAILINK_COMP_ARRAY(COMP_CPU("spi-RT5677AA:00")),
@@ -342,7 +350,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.init = bdw_rt5677_init,
-		SND_SOC_DAILINK_REG(dummy, be, dummy),
+		SND_SOC_DAILINK_REG(ssp0_port, be, platform),
 	},
 };
 
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index db7e1e87156d..b9c12e24c70b 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -164,6 +164,14 @@ SND_SOC_DAILINK_DEF(platform,
 SND_SOC_DAILINK_DEF(codec,
 	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1")));
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+SND_SOC_DAILINK_DEF(ssp0_port,
+	    DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
+#else
+SND_SOC_DAILINK_DEF(ssp0_port,
+	    DAILINK_COMP_ARRAY(COMP_DUMMY()));
+#endif
+
 /* broadwell digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 	/* Front End DAI links */
@@ -218,7 +226,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 		.ops = &broadwell_rt286_ops,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
-		SND_SOC_DAILINK_REG(dummy, codec, dummy),
+		SND_SOC_DAILINK_REG(ssp0_port, codec, platform),
 	},
 };
 
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 5873abb46441..33b13f3ca152 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -161,13 +161,13 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The ADSP will convert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = DUAL_CHANNEL;
+	chan->min = chan->max = DUAL_CHANNEL;
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
@@ -313,12 +313,12 @@ static const struct snd_soc_ops broxton_da7219_fe_ops = {
 static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 			struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
 	if (params_channels(params) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index eabf9d8468ae..067a97e7e6a8 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -206,13 +206,13 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 					SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 					SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The ADSP will covert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = 2;
+	chan->min = chan->max = 2;
 
 	/* set SSP5 to 24 bit */
 	snd_mask_none(fmt);
@@ -255,9 +255,9 @@ static const struct snd_pcm_hw_constraint_list constraints_rates = {
 static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 				struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
-	channels->min = channels->max = 4;
+	chan->min = chan->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 54e97455d7f6..0adc5a5e134a 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -234,14 +234,6 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime)
 	return 0;
 }
 
-static const struct snd_soc_pcm_stream byt_cht_es8316_dai_params = {
-	.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	.rate_min = 48000,
-	.rate_max = 48000,
-	.channels_min = 2,
-	.channels_max = 2,
-};
-
 static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 			    struct snd_pcm_hw_params *params)
 {
@@ -360,7 +352,10 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = {
 
 /* SoC card */
 static char codec_name[SND_ACPI_I2C_ID_LEN];
+#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
 static char long_name[50]; /* = "bytcht-es8316-*-spk-*-mic" */
+#endif
+static char components_string[32]; /* = "cfg-spk:* cfg-mic:* */
 
 static int byt_cht_es8316_suspend(struct snd_soc_card *card)
 {
@@ -573,11 +568,19 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 		}
 	}
 
-	/* register the soc card */
+	snprintf(components_string, sizeof(components_string),
+		 "cfg-spk:%s cfg-mic:%s",
+		 (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "1" : "2",
+		 mic_name[BYT_CHT_ES8316_MAP(quirk)]);
+	byt_cht_es8316_card.components = components_string;
+#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
 	snprintf(long_name, sizeof(long_name), "bytcht-es8316-%s-spk-%s-mic",
 		 (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "mono" : "stereo",
 		 mic_name[BYT_CHT_ES8316_MAP(quirk)]);
 	byt_cht_es8316_card.long_name = long_name;
+#endif
+
+	/* register the soc card */
 	snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv);
 
 	ret = devm_snd_soc_register_card(dev, &byt_cht_es8316_card);
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 243f683bc02a..6bd9ae813be2 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -933,14 +933,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
 	return 0;
 }
 
-static const struct snd_soc_pcm_stream byt_rt5640_dai_params = {
-	.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	.rate_min = 48000,
-	.rate_max = 48000,
-	.channels_min = 2,
-	.channels_max = 2,
-};
-
 static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 			    struct snd_pcm_hw_params *params)
 {
@@ -948,65 +940,43 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 			SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_interval *channels = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
-	int ret;
+	int ret, bits;
 
 	/* The DSP will covert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
 	channels->min = channels->max = 2;
 
 	if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) ||
-		(byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
-
+	    (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
 		/* set SSP0 to 16-bit */
 		params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
-
-		/*
-		 * Default mode for SSP configuration is TDM 4 slot, override config
-		 * with explicit setting to I2S 2ch 16-bit. The word length is set with
-		 * dai_set_tdm_slot() since there is no other API exposed
-		 */
-		ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
-					SND_SOC_DAIFMT_I2S     |
-					SND_SOC_DAIFMT_NB_NF   |
-					SND_SOC_DAIFMT_CBS_CFS
-			);
-		if (ret < 0) {
-			dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
-			return ret;
-		}
-
-		ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
-		if (ret < 0) {
-			dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
-			return ret;
-		}
-
+		bits = 16;
 	} else {
-
 		/* set SSP2 to 24-bit */
 		params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+		bits = 24;
+	}
 
-		/*
-		 * Default mode for SSP configuration is TDM 4 slot, override config
-		 * with explicit setting to I2S 2ch 24-bit. The word length is set with
-		 * dai_set_tdm_slot() since there is no other API exposed
-		 */
-		ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
-					SND_SOC_DAIFMT_I2S     |
-					SND_SOC_DAIFMT_NB_NF   |
-					SND_SOC_DAIFMT_CBS_CFS
-			);
-		if (ret < 0) {
-			dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
-			return ret;
-		}
+	/*
+	 * Default mode for SSP configuration is TDM 4 slot, override config
+	 * with explicit setting to I2S 2ch. The word length is set with
+	 * dai_set_tdm_slot() since there is no other API exposed
+	 */
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+				  SND_SOC_DAIFMT_I2S     |
+				  SND_SOC_DAIFMT_NB_NF   |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
+		return ret;
+	}
 
-		ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
-		if (ret < 0) {
-			dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
-			return ret;
-		}
+	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
+		return ret;
 	}
+
 	return 0;
 }
 
@@ -1084,9 +1054,10 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = {
 
 /* SoC card */
 static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN];
-static char byt_rt5640_codec_aif_name[12]; /*  = "rt5640-aif[1|2]" */
-static char byt_rt5640_cpu_dai_name[10]; /*  = "ssp[0|2]-port" */
+#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
 static char byt_rt5640_long_name[40]; /* = "bytcr-rt5640-*-spk-*-mic" */
+#endif
+static char byt_rt5640_components[32]; /* = "cfg-spk:* cfg-mic:*" */
 
 static int byt_rt5640_suspend(struct snd_soc_card *card)
 {
@@ -1266,28 +1237,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
 	log_quirks(&pdev->dev);
 
 	if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) ||
-	    (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
-
-		/* fixup codec aif name */
-		snprintf(byt_rt5640_codec_aif_name,
-			sizeof(byt_rt5640_codec_aif_name),
-			"%s", "rt5640-aif2");
-
-		byt_rt5640_dais[dai_index].codecs->dai_name =
-			byt_rt5640_codec_aif_name;
-	}
+	    (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2))
+		byt_rt5640_dais[dai_index].codecs->dai_name = "rt5640-aif2";
 
 	if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) ||
-	    (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
-
-		/* fixup cpu dai name name */
-		snprintf(byt_rt5640_cpu_dai_name,
-			sizeof(byt_rt5640_cpu_dai_name),
-			"%s", "ssp0-port");
-
-		byt_rt5640_dais[dai_index].cpus->dai_name =
-			byt_rt5640_cpu_dai_name;
-	}
+	    (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2))
+		byt_rt5640_dais[dai_index].cpus->dai_name = "ssp0-port";
 
 	if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) {
 		priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
@@ -1309,12 +1264,19 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
 		}
 	}
 
+	snprintf(byt_rt5640_components, sizeof(byt_rt5640_components),
+		 "cfg-spk:%s cfg-mic:%s",
+		 (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? "1" : "2",
+		 map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]);
+	byt_rt5640_card.components = byt_rt5640_components;
+#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
 	snprintf(byt_rt5640_long_name, sizeof(byt_rt5640_long_name),
 		 "bytcr-rt5640-%s-spk-%s-mic",
 		 (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ?
 			"mono" : "stereo",
 		 map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]);
 	byt_rt5640_card.long_name = byt_rt5640_long_name;
+#endif
 
 	/* override plaform name, if required */
 	platform_name = mach->mach_params.platform;
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 4606f6f582d6..5074bb53f98e 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -656,14 +656,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
 	return 0;
 }
 
-static const struct snd_soc_pcm_stream byt_rt5651_dai_params = {
-	.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	.rate_min = 48000,
-	.rate_max = 48000,
-	.channels_min = 2,
-	.channels_max = 2,
-};
-
 static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 			    struct snd_pcm_hw_params *params)
 {
@@ -795,9 +787,10 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
 
 /* SoC card */
 static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
-static char byt_rt5651_codec_aif_name[12]; /*  = "rt5651-aif[1|2]" */
-static char byt_rt5651_cpu_dai_name[10]; /*  = "ssp[0|2]-port" */
+#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
 static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */
+#endif
+static char byt_rt5651_components[50]; /* = "cfg-spk:* cfg-mic:*" */
 
 static int byt_rt5651_suspend(struct snd_soc_card *card)
 {
@@ -876,7 +869,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 	const char *platform_name;
 	struct acpi_device *adev;
 	struct device *codec_dev;
-	const char *hp_swapped;
 	bool is_bytcr = false;
 	int ret_val = 0;
 	int dai_index = 0;
@@ -996,10 +988,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 
 	if (byt_rt5651_gpios) {
 		devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios);
-		priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child(
-						&pdev->dev, "ext-amp-enable", 0,
-						codec_dev->fwnode,
-						GPIOD_OUT_LOW, "speaker-amp");
+		priv->ext_amp_gpio = devm_fwnode_gpiod_get(&pdev->dev,
+							   codec_dev->fwnode,
+							   "ext-amp-enable",
+							   GPIOD_OUT_LOW,
+							   "speaker-amp");
 		if (IS_ERR(priv->ext_amp_gpio)) {
 			ret_val = PTR_ERR(priv->ext_amp_gpio);
 			switch (ret_val) {
@@ -1015,10 +1008,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 				return ret_val;
 			}
 		}
-		priv->hp_detect = devm_fwnode_get_index_gpiod_from_child(
-						&pdev->dev, "hp-detect", 0,
-						codec_dev->fwnode,
-						GPIOD_IN, "hp-detect");
+		priv->hp_detect = devm_fwnode_gpiod_get(&pdev->dev,
+							codec_dev->fwnode,
+							"hp-detect",
+							GPIOD_IN,
+							"hp-detect");
 		if (IS_ERR(priv->hp_detect)) {
 			ret_val = PTR_ERR(priv->hp_detect);
 			switch (ret_val) {
@@ -1041,26 +1035,12 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 	log_quirks(&pdev->dev);
 
 	if ((byt_rt5651_quirk & BYT_RT5651_SSP2_AIF2) ||
-	    (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) {
-		/* fixup codec aif name */
-		snprintf(byt_rt5651_codec_aif_name,
-			sizeof(byt_rt5651_codec_aif_name),
-			"%s", "rt5651-aif2");
-
-		byt_rt5651_dais[dai_index].codecs->dai_name =
-			byt_rt5651_codec_aif_name;
-	}
+	    (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2))
+		byt_rt5651_dais[dai_index].codecs->dai_name = "rt5651-aif2";
 
 	if ((byt_rt5651_quirk & BYT_RT5651_SSP0_AIF1) ||
-	    (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) {
-		/* fixup cpu dai name name */
-		snprintf(byt_rt5651_cpu_dai_name,
-			sizeof(byt_rt5651_cpu_dai_name),
-			"%s", "ssp0-port");
-
-		byt_rt5651_dais[dai_index].cpus->dai_name =
-			byt_rt5651_cpu_dai_name;
-	}
+	    (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2))
+		byt_rt5651_dais[dai_index].cpus->dai_name = "ssp0-port";
 
 	if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) {
 		priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
@@ -1080,17 +1060,23 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 		}
 	}
 
-	if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED)
-		hp_swapped = "-hp-swapped";
-	else
-		hp_swapped = "";
-
+	snprintf(byt_rt5651_components, sizeof(byt_rt5651_components),
+		 "cfg-spk:%s cfg-mic:%s%s",
+		 (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? "1" : "2",
+		 mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)],
+		 (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ?
+			" cfg-hp:lrswap" : "");
+	byt_rt5651_card.components = byt_rt5651_components;
+#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
 	snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name),
 		 "bytcr-rt5651-%s-spk-%s-mic%s",
 		 (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ?
 			"mono" : "stereo",
-		 mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped);
+		 mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)],
+		 (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ?
+			"-hp-swapped" : "");
 	byt_rt5651_card.long_name = byt_rt5651_long_name;
+#endif
 
 	/* override plaform name, if required */
 	platform_name = mach->mach_params.platform;
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index c68a5b85a4a0..b5b016d493f1 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -515,8 +515,6 @@ static struct cht_acpi_card snd_soc_cards[] = {
 };
 
 static char cht_rt5645_codec_name[SND_ACPI_I2C_ID_LEN];
-static char cht_rt5645_codec_aif_name[12]; /*  = "rt5645-aif[1|2]" */
-static char cht_rt5645_cpu_dai_name[10]; /*  = "ssp[0|2]-port" */
 
 struct acpi_chan_package {   /* ACPICA seems to require 64 bit integers */
 	u64 aif_value;       /* 1: AIF1, 2: AIF2 */
@@ -641,28 +639,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
 	log_quirks(&pdev->dev);
 
 	if ((cht_rt5645_quirk & CHT_RT5645_SSP2_AIF2) ||
-		(cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) {
-
-		/* fixup codec aif name */
-		snprintf(cht_rt5645_codec_aif_name,
-			sizeof(cht_rt5645_codec_aif_name),
-			"%s", "rt5645-aif2");
-
-		cht_dailink[dai_index].codecs->dai_name =
-			cht_rt5645_codec_aif_name;
-	}
+	    (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2))
+		cht_dailink[dai_index].codecs->dai_name = "rt5645-aif2";
 
 	if ((cht_rt5645_quirk & CHT_RT5645_SSP0_AIF1) ||
-		(cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) {
-
-		/* fixup cpu dai name name */
-		snprintf(cht_rt5645_cpu_dai_name,
-			sizeof(cht_rt5645_cpu_dai_name),
-			"%s", "ssp0-port");
-
-		cht_dailink[dai_index].cpus->dai_name =
-			cht_rt5645_cpu_dai_name;
-	}
+	    (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2))
+		cht_dailink[dai_index].cpus->dai_name = "ssp0-port";
 
 	/* override plaform name, if required */
 	platform_name = mach->mach_params.platform;
diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c
index 5f1bf6d3800c..d6efc554898c 100644
--- a/sound/soc/intel/boards/cml_rt1011_rt5682.c
+++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c
@@ -405,19 +405,19 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = {
 
 static struct snd_soc_codec_conf rt1011_conf[] = {
 	{
-		.dev_name = "i2c-10EC1011:00",
+		.dlc = COMP_CODEC_CONF("i2c-10EC1011:00"),
 		.name_prefix = "WL",
 	},
 	{
-		.dev_name = "i2c-10EC1011:01",
+		.dlc = COMP_CODEC_CONF("i2c-10EC1011:01"),
 		.name_prefix = "WR",
 	},
 	{
-		.dev_name = "i2c-10EC1011:02",
+		.dlc = COMP_CODEC_CONF("i2c-10EC1011:02"),
 		.name_prefix = "TL",
 	},
 	{
-		.dev_name = "i2c-10EC1011:03",
+		.dlc = COMP_CODEC_CONF("i2c-10EC1011:03"),
 		.name_prefix = "TR",
 	},
 };
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index b36264d1d1cd..4a6d117ea7af 100644
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -118,13 +118,13 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The ADSP will convert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = DUAL_CHANNEL;
+	chan->min = chan->max = DUAL_CHANNEL;
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
@@ -249,16 +249,6 @@ static const struct snd_pcm_hw_constraint_list constraints_rates = {
 	.mask = 0,
 };
 
-static const unsigned int channels[] = {
-	DUAL_CHANNEL,
-};
-
-static const struct snd_pcm_hw_constraint_list constraints_channels = {
-	.count = ARRAY_SIZE(channels),
-	.list = channels,
-	.mask = 0,
-};
-
 static unsigned int channels_quad[] = {
 	QUAD_CHANNEL,
 };
@@ -272,13 +262,13 @@ static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
 static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 		struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 				SNDRV_PCM_HW_PARAM_CHANNELS);
 
 	/*
 	 * set BE channel constraint as user FE channels
 	 */
-	channels->min = channels->max = 4;
+	chan->min = chan->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c
index ed36b68d6705..9179f07f9ee4 100644
--- a/sound/soc/intel/boards/hda_dsp_common.c
+++ b/sound/soc/intel/boards/hda_dsp_common.c
@@ -63,7 +63,7 @@ int hda_dsp_hdmi_build_controls(struct snd_soc_card *card,
 				"%s: mapping HDMI converter %d to PCM %d (%p)\n",
 				__func__, i, hpcm->device, spcm);
 		} else {
-			hpcm->pcm = 0;
+			hpcm->pcm = NULL;
 			hpcm->device = SNDRV_PCM_INVALID_DEVICE;
 			dev_warn(card->dev,
 				 "%s: no PCM in topology for HDMI converter %d\n\n",
diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c
index 537a88932bb6..bc7f9a9ce9af 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98357a.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c
@@ -141,13 +141,13 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The ADSP will convert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = DUAL_CHANNEL;
+	chan->min = chan->max = DUAL_CHANNEL;
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
@@ -305,7 +305,7 @@ static const struct snd_soc_ops kabylake_da7219_fe_ops = {
 static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 		struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 				SNDRV_PCM_HW_PARAM_CHANNELS);
 
 	/*
@@ -313,9 +313,9 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 	 */
 
 	if (params_channels(params) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
@@ -336,19 +336,6 @@ static struct snd_soc_ops kabylake_dmic_ops = {
 	.startup = kabylake_dmic_startup,
 };
 
-static const unsigned int rates_16000[] = {
-	16000,
-};
-
-static const struct snd_pcm_hw_constraint_list constraints_16000 = {
-	.count = ARRAY_SIZE(rates_16000),
-	.list  = rates_16000,
-};
-
-static const unsigned int ch_mono[] = {
-	1,
-};
-
 SND_SOC_DAILINK_DEF(dummy,
 	DAILINK_COMP_ARRAY(COMP_DUMMY()));
 
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 829f95fc4179..7a13e9b35187 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -176,7 +176,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *runtime = substream->private_data;
-	int ret = 0, j;
+	int ret, j;
 
 	for (j = 0; j < runtime->num_codecs; j++) {
 		struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
@@ -279,7 +279,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	struct snd_soc_dpcm *dpcm = container_of(
@@ -298,7 +298,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 		!strcmp(rtd->card->name, "kblmax98373")) {
 		/* The ADSP will convert the FE rate to 48k, stereo */
 		rate->min = rate->max = 48000;
-		channels->min = channels->max = DUAL_CHANNEL;
+		chan->min = chan->max = DUAL_CHANNEL;
 
 		/* set SSP to 24 bit */
 		snd_mask_none(fmt);
@@ -313,7 +313,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
 	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	}
@@ -341,6 +341,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
 			kabylake_ssp1_map,
 			ARRAY_SIZE(kabylake_ssp1_map));
 
+	if (ret)
+		return ret;
+
 	/*
 	 * Headset buttons map to the google Reference headset.
 	 * These can be configured by userspace.
@@ -491,7 +494,7 @@ static const struct snd_soc_ops kabylake_da7219_fe_ops = {
 static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 		struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 				SNDRV_PCM_HW_PARAM_CHANNELS);
 
 	/*
@@ -499,9 +502,9 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 	 */
 
 	if (params_channels(params) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
@@ -571,12 +574,12 @@ static struct snd_soc_ops skylake_refcap_ops = {
 static struct snd_soc_codec_conf max98927_codec_conf[] = {
 
 	{
-		.dev_name = MAX98927_DEV0_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98927_DEV0_NAME),
 		.name_prefix = "Right",
 	},
 
 	{
-		.dev_name = MAX98927_DEV1_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98927_DEV1_NAME),
 		.name_prefix = "Left",
 	},
 };
@@ -584,12 +587,12 @@ static struct snd_soc_codec_conf max98927_codec_conf[] = {
 static struct snd_soc_codec_conf max98373_codec_conf[] = {
 
 	{
-		.dev_name = MAX98373_DEV0_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME),
 		.name_prefix = "Right",
 	},
 
 	{
-		.dev_name = MAX98373_DEV1_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME),
 		.name_prefix = "Left",
 	},
 };
@@ -1092,7 +1095,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
 	struct kbl_codec_private *ctx;
 	struct snd_soc_dai_link *kbl_dai_link;
 	struct snd_soc_dai_link_component **codecs;
-	int i = 0;
+	int i;
 
 	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
 	if (!ctx)
diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c
index 74fe1f3a5479..e23dea9ab79a 100644
--- a/sound/soc/intel/boards/kbl_rt5660.c
+++ b/sound/soc/intel/boards/kbl_rt5660.c
@@ -138,13 +138,13 @@ static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The ADSP will convert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = DUAL_CHANNEL;
+	chan->min = chan->max = DUAL_CHANNEL;
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 7cefda341fbf..d8f2ff7139a9 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -229,11 +229,11 @@ static const struct snd_soc_dapm_route kabylake_5663_map[] = {
 
 static struct snd_soc_codec_conf max98927_codec_conf[] = {
 	{
-		.dev_name = MAXIM_DEV0_NAME,
+		.dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME),
 		.name_prefix = "Right",
 	},
 	{
-		.dev_name = MAXIM_DEV1_NAME,
+		.dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME),
 		.name_prefix = "Left",
 	},
 };
@@ -398,7 +398,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	struct snd_soc_dpcm *dpcm = container_of(
@@ -413,7 +413,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
 	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	}
@@ -454,13 +454,13 @@ static struct snd_soc_ops kabylake_rt5663_ops = {
 static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 		struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 				SNDRV_PCM_HW_PARAM_CHANNELS);
 
 	if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 3e5f6bead229..96c814f36458 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -193,11 +193,11 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 
 static struct snd_soc_codec_conf max98927_codec_conf[] = {
 	{
-		.dev_name = MAXIM_DEV0_NAME,
+		.dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME),
 		.name_prefix = "Right",
 	},
 	{
-		.dev_name = MAXIM_DEV1_NAME,
+		.dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME),
 		.name_prefix = "Left",
 	},
 };
@@ -333,7 +333,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	struct snd_soc_dpcm *dpcm = container_of(
@@ -348,15 +348,15 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
 	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	} else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) {
 		if (params_channels(params) == 2 ||
 				DMIC_CH(dmic_constraints) == 2)
-			channels->min = channels->max = 2;
+			chan->min = chan->max = 2;
 		else
-			channels->min = channels->max = 4;
+			chan->min = chan->max = 4;
 	}
 	/*
 	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
@@ -761,7 +761,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
 {
 	struct kbl_codec_private *ctx;
 	struct snd_soc_acpi_mach *mach;
-	int ret = 0;
+	int ret;
 
 	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
 	if (!ctx)
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 4e45901e3a2f..11eaee9ae41f 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -100,6 +100,8 @@ static struct snd_soc_card hda_soc_card = {
 	.late_probe = skl_hda_card_late_probe,
 };
 
+static char hda_soc_components[30];
+
 #define IDISP_DAI_COUNT		3
 #define HDAC_DAI_COUNT		2
 #define DMIC_DAI_COUNT		2
@@ -183,6 +185,12 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
 	hda_soc_card.dev = &pdev->dev;
 	snd_soc_card_set_drvdata(&hda_soc_card, ctx);
 
+	if (mach->mach_params.dmic_num > 0) {
+		snprintf(hda_soc_components, sizeof(hda_soc_components),
+				"cfg-dmics:%d", mach->mach_params.dmic_num);
+		hda_soc_card.components = hda_soc_components;
+	}
+
 	return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card);
 }
 
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 3ce8efbeed12..e6de3b28d840 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -139,13 +139,13 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The ADSP will covert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = 2;
+	chan->min = chan->max = 2;
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
@@ -315,13 +315,13 @@ static const struct snd_soc_ops skylake_nau8825_ops = {
 static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 		struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 				SNDRV_PCM_HW_PARAM_CHANNELS);
 
 	if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index 1a7ac8bdf543..c99c8b23e509 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -147,11 +147,11 @@ static const struct snd_soc_dapm_route skylake_map[] = {
 
 static struct snd_soc_codec_conf ssm4567_codec_conf[] = {
 	{
-		.dev_name = "i2c-INT343B:00",
+		.dlc = COMP_CODEC_CONF("i2c-INT343B:00"),
 		.name_prefix = "Left",
 	},
 	{
-		.dev_name = "i2c-INT343B:01",
+		.dlc = COMP_CODEC_CONF("i2c-INT343B:01"),
 		.name_prefix = "Right",
 	},
 };
@@ -317,13 +317,13 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The ADSP will covert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = 2;
+	chan->min = chan->max = 2;
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
@@ -334,12 +334,12 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 			struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
 	if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 231349a47cc9..a9aec66a2351 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -211,13 +211,13 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	/* The output is 48KHz, stereo, 16bits */
 	rate->min = rate->max = 48000;
-	channels->min = channels->max = 2;
+	chan->min = chan->max = 2;
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
@@ -247,12 +247,12 @@ static const struct snd_soc_ops skylake_rt286_ops = {
 static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 				struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
 	if (params_channels(params) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c
new file mode 100644
index 000000000000..8f44f13d2848
--- /dev/null
+++ b/sound/soc/intel/boards/sof_da7219_max98373.c
@@ -0,0 +1,371 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2019 Intel Corporation.
+
+/*
+ * Intel SOF Machine driver for DA7219 + MAX98373 codec
+ */
+
+#include <linux/input.h>
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <linux/platform_device.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "../../codecs/da7219.h"
+#include "../../codecs/da7219-aad.h"
+#include "hda_dsp_common.h"
+
+#define DIALOG_CODEC_DAI	"da7219-hifi"
+#define MAX98373_CODEC_DAI	"max98373-aif1"
+#define MAXIM_DEV0_NAME		"i2c-MX98373:00"
+#define MAXIM_DEV1_NAME		"i2c-MX98373:01"
+
+struct hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct card_private {
+	struct snd_soc_jack headset;
+	struct list_head hdmi_pcm_list;
+	struct snd_soc_jack hdmi[3];
+};
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+				  struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret = 0;
+
+	codec_dai = snd_soc_card_get_codec_dai(card, DIALOG_CODEC_DAI);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+		return -EIO;
+	}
+
+	if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK,
+					  0, 0);
+		if (ret)
+			dev_err(card->dev, "failed to stop PLL: %d\n", ret);
+	} else if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM,
+					  0, DA7219_PLL_FREQ_OUT_98304);
+		if (ret)
+			dev_err(card->dev, "failed to start PLL: %d\n", ret);
+	}
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const struct snd_soc_dapm_widget widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			    platform_clock_control, SND_SOC_DAPM_POST_PMD |
+			    SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{ "Headphone Jack", NULL, "HPL" },
+	{ "Headphone Jack", NULL, "HPR" },
+
+	{ "Left Spk", NULL, "Left BE_OUT" },
+	{ "Right Spk", NULL, "Right BE_OUT" },
+
+	{ "MIC", NULL, "Headset Mic" },
+
+	{ "Headphone Jack", NULL, "Platform Clock" },
+	{ "Headset Mic", NULL, "Platform Clock" },
+};
+
+static struct snd_soc_jack headset;
+
+static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_jack *jack;
+	int ret;
+
+	/* Configure sysclk for codec */
+	ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24000000,
+				     SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	/*
+	 * Headset buttons map to the google Reference headset.
+	 * These can be configured by userspace.
+	 */
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+				    SND_JACK_HEADSET | SND_JACK_BTN_0 |
+				    SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+				    SND_JACK_BTN_3 | SND_JACK_LINEOUT,
+				    &headset, NULL, 0);
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+		return ret;
+	}
+
+	jack = &headset;
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+	da7219_aad_jack_det(component, jack);
+
+	return ret;
+}
+
+static int ssp1_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *runtime = substream->private_data;
+	int ret, j;
+
+	for (j = 0; j < runtime->num_codecs; j++) {
+		struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
+			/* vmon_slot_no = 0 imon_slot_no = 1 for TX slots */
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 3, 4, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
+			/* vmon_slot_no = 2 imon_slot_no = 3 for TX slots */
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC, 3, 4, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops ssp1_ops = {
+	.hw_params = ssp1_hw_params,
+};
+
+static struct snd_soc_codec_conf max98373_codec_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME),
+		.name_prefix = "Right",
+	},
+	{
+		.dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME),
+		.name_prefix = "Left",
+	},
+};
+
+static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = dai->id;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+static int card_late_probe(struct snd_soc_card *card)
+{
+	struct card_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_acpi_mach *mach = (card->dev)->platform_data;
+	struct hdmi_pcm *pcm;
+
+	if (mach->mach_params.common_hdmi_codec_drv) {
+		pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm,
+				       head);
+		return hda_dsp_hdmi_build_controls(card,
+						   pcm->codec_dai->component);
+	}
+
+	return -EINVAL;
+}
+
+SND_SOC_DAILINK_DEF(ssp0_pin,
+	DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin")));
+SND_SOC_DAILINK_DEF(ssp0_codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", DIALOG_CODEC_DAI)));
+
+SND_SOC_DAILINK_DEF(ssp1_pin,
+	DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin")));
+SND_SOC_DAILINK_DEF(ssp1_amps,
+	DAILINK_COMP_ARRAY(
+	/* Left */	COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI),
+	/* Right */	COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI)));
+
+SND_SOC_DAILINK_DEF(dmic_pin,
+	DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
+SND_SOC_DAILINK_DEF(dmic_codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi")));
+
+SND_SOC_DAILINK_DEF(idisp1_pin,
+	DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")));
+SND_SOC_DAILINK_DEF(idisp1_codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1")));
+
+SND_SOC_DAILINK_DEF(idisp2_pin,
+	DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")));
+SND_SOC_DAILINK_DEF(idisp2_codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2")));
+
+SND_SOC_DAILINK_DEF(idisp3_pin,
+	DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")));
+SND_SOC_DAILINK_DEF(idisp3_codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3")));
+
+SND_SOC_DAILINK_DEF(platform, /* subject to be overridden during probe */
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3")));
+
+static struct snd_soc_dai_link dais[] = {
+	/* Back End DAI links */
+	{
+		.name = "SSP1-Codec",
+		.id = 0,
+		.ignore_pmdown_time = 1,
+		.no_pcm = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1, /* IV feedback */
+		.ops = &ssp1_ops,
+		SND_SOC_DAILINK_REG(ssp1_pin, ssp1_amps, platform),
+	},
+	{
+		.name = "SSP0-Codec",
+		.id = 1,
+		.no_pcm = 1,
+		.init = da7219_codec_init,
+		.ignore_pmdown_time = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform),
+	},
+	{
+		.name = "dmic01",
+		.id = 2,
+		.ignore_suspend = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+		SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform),
+	},
+	{
+		.name = "iDisp1",
+		.id = 3,
+		.init = hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+		SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform),
+	},
+	{
+		.name = "iDisp2",
+		.id = 4,
+		.init = hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+		SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform),
+	},
+	{
+		.name = "iDisp3",
+		.id = 5,
+		.init = hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+		SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform),
+	},
+};
+
+static struct snd_soc_card card_da7219_m98373 = {
+	.name = "da7219max",
+	.owner = THIS_MODULE,
+	.dai_link = dais,
+	.num_links = ARRAY_SIZE(dais),
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
+	.dapm_widgets = widgets,
+	.num_dapm_widgets = ARRAY_SIZE(widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.codec_conf = max98373_codec_conf,
+	.num_configs = ARRAY_SIZE(max98373_codec_conf),
+	.fully_routed = true,
+	.late_probe = card_late_probe,
+};
+
+static int audio_probe(struct platform_device *pdev)
+{
+	static struct snd_soc_card *card;
+	struct snd_soc_acpi_mach *mach;
+	struct card_private *ctx;
+	int ret;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+	card = (struct snd_soc_card *)pdev->id_entry->driver_data;
+	card->dev = &pdev->dev;
+
+	mach = (&pdev->dev)->platform_data;
+	ret = snd_soc_fixup_dai_links_platform_name(card,
+						    mach->mach_params.platform);
+	if (ret)
+		return ret;
+
+	snd_soc_card_set_drvdata(card, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev, card);
+}
+
+static const struct platform_device_id board_ids[] = {
+	{
+		.name = "sof_da7219_max98373",
+		.driver_data = (kernel_ulong_t)&card_da7219_m98373,
+	},
+	{ }
+};
+
+static struct platform_driver audio = {
+	.probe = audio_probe,
+	.driver = {
+		.name = "sof_da7219_max98373",
+		.pm = &snd_soc_pm_ops,
+	},
+	.id_table = board_ids,
+};
+module_platform_driver(audio)
+
+/* Module information */
+MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver");
+MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_da7219_max98373");
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 751b8ea6ae1f..8a13231dee15 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -35,6 +35,10 @@
 #define SOF_RT5682_SSP_AMP(quirk)	\
 	(((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK)
 #define SOF_RT5682_MCLK_BYTCHT_EN		BIT(9)
+#define SOF_RT5682_NUM_HDMIDEV_SHIFT		10
+#define SOF_RT5682_NUM_HDMIDEV_MASK		(GENMASK(12, 10))
+#define SOF_RT5682_NUM_HDMIDEV(quirk)	\
+	((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK)
 
 /* Default: MCLK on, MCLK 19.2M, SSP0  */
 static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
@@ -594,6 +598,19 @@ static int sof_audio_probe(struct platform_device *pdev)
 	if (!ctx)
 		return -ENOMEM;
 
+	if (pdev->id_entry && pdev->id_entry->driver_data)
+		sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data;
+
+	dmi_check_system(sof_rt5682_quirk_table);
+
+	mach = (&pdev->dev)->platform_data;
+
+	/* A speaker amp might not be present when the quirk claims one is.
+	 * Detect this via whether the machine driver match includes quirk_data.
+	 */
+	if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data)
+		sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT;
+
 	if (soc_intel_is_byt() || soc_intel_is_cht()) {
 		is_legacy_cpu = 1;
 		dmic_be_num = 0;
@@ -604,11 +621,13 @@ static int sof_audio_probe(struct platform_device *pdev)
 						SOF_RT5682_SSP_CODEC(2);
 	} else {
 		dmic_be_num = 2;
-		hdmi_num = 3;
+		hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >>
+			 SOF_RT5682_NUM_HDMIDEV_SHIFT;
+		/* default number of HDMI DAI's */
+		if (!hdmi_num)
+			hdmi_num = 3;
 	}
 
-	dmi_check_system(sof_rt5682_quirk_table);
-
 	/* need to get main clock from pmc */
 	if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
 		ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
@@ -652,7 +671,6 @@ static int sof_audio_probe(struct platform_device *pdev)
 	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
 
 	sof_audio_card_rt5682.dev = &pdev->dev;
-	mach = (&pdev->dev)->platform_data;
 
 	/* set platform name for each dailink */
 	ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682,
@@ -683,6 +701,21 @@ static int sof_rt5682_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct platform_device_id board_ids[] = {
+	{
+		.name = "sof_rt5682",
+	},
+	{
+		.name = "tgl_max98357a_rt5682",
+		.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+					SOF_RT5682_SSP_CODEC(0) |
+					SOF_SPEAKER_AMP_PRESENT |
+					SOF_RT5682_SSP_AMP(1) |
+					SOF_RT5682_NUM_HDMIDEV(4)),
+	},
+	{ }
+};
+
 static struct platform_driver sof_audio = {
 	.probe = sof_audio_probe,
 	.remove = sof_rt5682_remove,
@@ -690,6 +723,7 @@ static struct platform_driver sof_audio = {
 		.name = "sof_rt5682",
 		.pm = &snd_soc_pm_ops,
 	},
+	.id_table = board_ids,
 };
 module_platform_driver(sof_audio)
 
@@ -699,3 +733,4 @@ MODULE_AUTHOR("Bard Liao <bard.liao@intel.com>");
 MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>");
 MODULE_LICENSE("GPL v2");
 MODULE_ALIAS("platform:sof_rt5682");
+MODULE_ALIAS("platform:tgl_max98357a_rt5682");