Merge tag 'asoc-v5.3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v5.3

This is a very big update, mainly thanks to Morimoto-san's refactoring
work and some fairly large new drivers.

 - Lots more work on moving towards a component based framework from
   Morimoto-san.
 - Support for force disconnecting muxes from Jerome Brunet.
 - New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
   CX2072X, Realtek RT1011 and RT1308.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

23 files changed, 259 insertions, 365 deletions

diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index ee7c202c69b7..2197f3e1eaed 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -1,3 +1,4 @@
+# SPDX-License-Identifier: GPL-2.0-only
 menu "Audio support for Texas Instruments SoCs"
 depends on DMA_OMAP || TI_EDMA || COMPILE_TEST
 
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index b9611db14c86..dee8fc70a64f 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * ams-delta.c  --  SoC audio for Amstrad E3 (Delta) videophone
  *
@@ -5,21 +6,6 @@
  *
  * Initially based on sound/soc/omap/osk5912.x
  * Copyright (C) 2008 Mistral Solutions
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/gpio/consumer.h>
@@ -518,17 +504,19 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 }
 
 /* DAI glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(cx20442,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("cx20442-codec", "cx20442-voice")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.1")));
+
 static struct snd_soc_dai_link ams_delta_dai_link = {
 	.name = "CX20442",
 	.stream_name = "CX20442",
-	.cpu_dai_name = "omap-mcbsp.1",
-	.codec_dai_name = "cx20442-voice",
 	.init = ams_delta_cx20442_init,
-	.platform_name = "omap-mcbsp.1",
-	.codec_name = "cx20442-codec",
 	.ops = &ams_delta_ops,
 	.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
 		   SND_SOC_DAIFMT_CBM_CFM,
+	SND_SOC_DAILINK_REG(cx20442),
 };
 
 /* Audio card driver */
diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
index 4869d6311510..bfd8d1a03ba7 100644
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -1,12 +1,9 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * ASoC driver for TI DAVINCI EVM platform
  *
  * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
  * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #include <linux/module.h>
@@ -143,103 +140,127 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 }
 
 /* davinci-evm digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(dm6446,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-001b",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp")));
+
 static struct snd_soc_dai_link dm6446_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-mcbsp",
-	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.1-001b",
-	.platform_name = "davinci-mcbsp",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
 	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(dm6446),
 };
 
+SND_SOC_DAILINK_DEFS(dm355,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-001b",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp.1")));
+
 static struct snd_soc_dai_link dm355_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-mcbsp.1",
-	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.1-001b",
-	.platform_name = "davinci-mcbsp.1",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
 	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(dm355),
 };
 
+#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC
+SND_SOC_DAILINK_DEFS(dm365,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp")));
+#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC)
+SND_SOC_DAILINK_DEFS(dm365,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-vcif")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("cq93vc-codec", "cq93vc-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-vcif")));
+#endif
+
 static struct snd_soc_dai_link dm365_evm_dai = {
 #ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-mcbsp",
-	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.1-0018",
-	.platform_name = "davinci-mcbsp",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
 	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(dm365),
 #elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC)
 	.name = "Voice Codec - CQ93VC",
 	.stream_name = "CQ93",
-	.cpu_dai_name = "davinci-vcif",
-	.codec_dai_name = "cq93vc-hifi",
-	.codec_name = "cq93vc-codec",
-	.platform_name = "davinci-vcif",
+	SND_SOC_DAILINK_REG(dm365),
 #endif
 };
 
+SND_SOC_DAILINK_DEFS(dm6467_aic3x,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.0")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.0-001a",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.0")));
+
+SND_SOC_DAILINK_DEFS(dm6467_spdif,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("spdif_dit", "dit-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.1")));
+
 static struct snd_soc_dai_link dm6467_evm_dai[] = {
 	{
 		.name = "TLV320AIC3X",
 		.stream_name = "AIC3X",
-		.cpu_dai_name= "davinci-mcasp.0",
-		.codec_dai_name = "tlv320aic3x-hifi",
-		.platform_name = "davinci-mcasp.0",
-		.codec_name = "tlv320aic3x-codec.0-001a",
 		.init = evm_aic3x_init,
 		.ops = &evm_ops,
 		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 			   SND_SOC_DAIFMT_IB_NF,
+		SND_SOC_DAILINK_REG(dm6467_aic3x),
 	},
 	{
 		.name = "McASP",
 		.stream_name = "spdif",
-		.cpu_dai_name= "davinci-mcasp.1",
-		.codec_dai_name = "dit-hifi",
-		.codec_name = "spdif_dit",
-		.platform_name = "davinci-mcasp.1",
 		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 			   SND_SOC_DAIFMT_IB_NF,
+		SND_SOC_DAILINK_REG(dm6467_spdif),
 	},
 };
 
+SND_SOC_DAILINK_DEFS(da830,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.1")));
+
 static struct snd_soc_dai_link da830_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-mcasp.1",
-	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.1-0018",
-	.platform_name = "davinci-mcasp.1",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
 	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(da830),
 };
 
+SND_SOC_DAILINK_DEFS(da850,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.0")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.0")));
+
 static struct snd_soc_dai_link da850_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name= "davinci-mcasp.0",
-	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.1-0018",
-	.platform_name = "davinci-mcasp.0",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
 	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(da850),
 };
 
 /* davinci dm6446 evm audio machine driver */
@@ -330,14 +351,19 @@ static struct snd_soc_card da850_snd_soc_card = {
  * The struct is used as place holder. It will be completely
  * filled with data from dt node.
  */
+SND_SOC_DAILINK_DEFS(evm,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
 static struct snd_soc_dai_link evm_dai_tlv320aic3x = {
 	.name		= "TLV320AIC3X",
 	.stream_name	= "AIC3X",
-	.codec_dai_name	= "tlv320aic3x-hifi",
 	.ops            = &evm_ops,
 	.init           = evm_aic3x_init,
 	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
 		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(evm),
 };
 
 static const struct of_device_id davinci_evm_dt_ids[] = {
@@ -374,15 +400,15 @@ static int davinci_evm_probe(struct platform_device *pdev)
 
 	evm_soc_card.dai_link = dai;
 
-	dai->codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0);
-	if (!dai->codec_of_node)
+	dai->codecs->of_node = of_parse_phandle(np, "ti,audio-codec", 0);
+	if (!dai->codecs->of_node)
 		return -EINVAL;
 
-	dai->cpu_of_node = of_parse_phandle(np, "ti,mcasp-controller", 0);
-	if (!dai->cpu_of_node)
+	dai->cpus->of_node = of_parse_phandle(np, "ti,mcasp-controller", 0);
+	if (!dai->cpus->of_node)
 		return -EINVAL;
 
-	dai->platform_of_node = dai->cpu_of_node;
+	dai->platforms->of_node = dai->cpus->of_node;
 
 	evm_soc_card.dev = &pdev->dev;
 	ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model");
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index a3206e65e5e5..92c1bdc69086 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
  *
@@ -7,10 +8,6 @@
  * DT support	(c) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
  *		based on davinci-mcasp.c DT support
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
  * TODO:
  * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers
  */
diff --git a/sound/soc/ti/davinci-i2s.h b/sound/soc/ti/davinci-i2s.h
index 48dac3e2521a..88d4df1d16de 100644
--- a/sound/soc/ti/davinci-i2s.h
+++ b/sound/soc/ti/davinci-i2s.h
@@ -1,12 +1,9 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
 /*
  * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
  *
  * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
  * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef _DAVINCI_I2S_H
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 9fbc759fdefe..ac59b509ead5 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * ALSA SoC McASP Audio Layer for TI DAVINCI processor
  *
@@ -9,10 +10,6 @@
  *
  * Copyright:   (C) 2009 MontaVista Software, Inc., <source@mvista.com>
  * Copyright:   (C) 2009  Texas Instruments, India
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #include <linux/init.h>
@@ -100,6 +97,7 @@ struct davinci_mcasp {
 
 	int	sysclk_freq;
 	bool	bclk_master;
+	u32	auxclk_fs_ratio;
 
 	unsigned long pdir; /* Pin direction bitfield */
 
@@ -847,14 +845,15 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 					rx_ser < max_active_serializers) {
 			clear_bit(PIN_BIT_AXR(i), &mcasp->pdir);
 			rx_ser++;
-		} else if (mcasp->serial_dir[i] == INACTIVE_MODE) {
+		} else {
+			/* Inactive or unused pin, set it to inactive */
 			mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
 				       SRMOD_INACTIVE, SRMOD_MASK);
-			clear_bit(PIN_BIT_AXR(i), &mcasp->pdir);
-		} else if (mcasp->serial_dir[i] == TX_MODE) {
-			/* Unused TX pins, clear PDIR  */
-			mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
-				       mcasp->dismod, DISMOD_MASK);
+			/* If unused, set DISMOD for the pin */
+			if (mcasp->serial_dir[i] != INACTIVE_MODE)
+				mcasp_mod_bits(mcasp,
+					       DAVINCI_MCASP_XRSRCTL_REG(i),
+					       mcasp->dismod, DISMOD_MASK);
 			clear_bit(PIN_BIT_AXR(i), &mcasp->pdir);
 		}
 	}
@@ -944,14 +943,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
 		active_slots = hweight32(mcasp->tdm_mask[stream]);
 		active_serializers = (channels + active_slots - 1) /
 			active_slots;
-		if (active_serializers == 1) {
+		if (active_serializers == 1)
 			active_slots = channels;
-			for (i = 0; i < total_slots; i++) {
-				if ((1 << i) & mcasp->tdm_mask[stream]) {
-					mask |= (1 << i);
-					if (--active_slots <= 0)
-						break;
-				}
+		for (i = 0; i < total_slots; i++) {
+			if ((1 << i) & mcasp->tdm_mask[stream]) {
+				mask |= (1 << i);
+				if (--active_slots <= 0)
+					break;
 			}
 		}
 	} else {
@@ -964,6 +962,7 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
 		for (i = 0; i < active_slots; i++)
 			mask |= (1 << i);
 	}
+
 	mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
 
 	if (!mcasp->dat_port)
@@ -1064,13 +1063,13 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp,
 }
 
 static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
+				      unsigned int sysclk_freq,
 				      unsigned int bclk_freq, bool set)
 {
-	int error_ppm;
-	unsigned int sysclk_freq = mcasp->sysclk_freq;
 	u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG);
 	int div = sysclk_freq / bclk_freq;
 	int rem = sysclk_freq % bclk_freq;
+	int error_ppm;
 	int aux_div = 1;
 
 	if (div > (ACLKXDIV_MASK + 1)) {
@@ -1175,7 +1174,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		if (mcasp->slot_width)
 			sbits = mcasp->slot_width;
 
-		davinci_mcasp_calc_clk_div(mcasp, rate * sbits * slots, true);
+		davinci_mcasp_calc_clk_div(mcasp, mcasp->sysclk_freq,
+					   rate * sbits * slots, true);
 	}
 
 	ret = mcasp_common_hw_param(mcasp, substream->stream,
@@ -1282,12 +1282,19 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
 
 	for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) {
 		if (snd_interval_test(ri, davinci_mcasp_dai_rates[i])) {
-			uint bclk_freq = sbits*slots*
-				davinci_mcasp_dai_rates[i];
+			uint bclk_freq = sbits * slots *
+					 davinci_mcasp_dai_rates[i];
+			unsigned int sysclk_freq;
 			int ppm;
 
-			ppm = davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq,
-							 false);
+			if (rd->mcasp->auxclk_fs_ratio)
+				sysclk_freq =  davinci_mcasp_dai_rates[i] *
+					       rd->mcasp->auxclk_fs_ratio;
+			else
+				sysclk_freq = rd->mcasp->sysclk_freq;
+
+			ppm = davinci_mcasp_calc_clk_div(rd->mcasp, sysclk_freq,
+							 bclk_freq, false);
 			if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
 				if (range.empty) {
 					range.min = davinci_mcasp_dai_rates[i];
@@ -1321,12 +1328,19 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
 	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
 		if (snd_mask_test(fmt, i)) {
 			uint sbits = snd_pcm_format_width(i);
+			unsigned int sysclk_freq;
 			int ppm;
 
+			if (rd->mcasp->auxclk_fs_ratio)
+				sysclk_freq =  rate *
+					       rd->mcasp->auxclk_fs_ratio;
+			else
+				sysclk_freq = rd->mcasp->sysclk_freq;
+
 			if (rd->mcasp->slot_width)
 				sbits = rd->mcasp->slot_width;
 
-			ppm = davinci_mcasp_calc_clk_div(rd->mcasp,
+			ppm = davinci_mcasp_calc_clk_div(rd->mcasp, sysclk_freq,
 							 sbits * slots * rate,
 							 false);
 			if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
@@ -1991,6 +2005,22 @@ static inline int davinci_mcasp_init_gpiochip(struct davinci_mcasp *mcasp)
 }
 #endif /* CONFIG_GPIOLIB */
 
+static int davinci_mcasp_get_dt_params(struct davinci_mcasp *mcasp)
+{
+	struct device_node *np = mcasp->dev->of_node;
+	int ret;
+	u32 val;
+
+	if (!np)
+		return 0;
+
+	ret = of_property_read_u32(np, "auxclk-fs-ratio", &val);
+	if (ret >= 0)
+		mcasp->auxclk_fs_ratio = val;
+
+	return 0;
+}
+
 static int davinci_mcasp_probe(struct platform_device *pdev)
 {
 	struct snd_dmaengine_dai_dma_data *dma_data;
@@ -2224,6 +2254,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (ret)
 		goto err;
 
+	ret = davinci_mcasp_get_dt_params(mcasp);
+	if (ret)
+		return -EINVAL;
+
 	ret = devm_snd_soc_register_component(&pdev->dev,
 					&davinci_mcasp_component,
 					&davinci_mcasp_dai[pdata->op_mode], 1);
@@ -2237,7 +2271,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		ret = edma_pcm_platform_register(&pdev->dev);
 		break;
 	case PCM_SDMA:
-		ret = sdma_pcm_platform_register(&pdev->dev, NULL, NULL);
+		ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
 		break;
 	default:
 		dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret);
diff --git a/sound/soc/ti/davinci-mcasp.h b/sound/soc/ti/davinci-mcasp.h
index 5e4060d8fe56..bc705d6ca48b 100644
--- a/sound/soc/ti/davinci-mcasp.h
+++ b/sound/soc/ti/davinci-mcasp.h
@@ -1,3 +1,4 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
 /*
  * ALSA SoC McASP Audio Layer for TI DAVINCI processor
  *
@@ -9,10 +10,6 @@
  *
  * Copyright:   (C) 2009 MontaVista Software, Inc., <source@mvista.com>
  * Copyright:   (C) 2009  Texas Instruments, India
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef DAVINCI_MCASP_H
diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c
index 5415b72393fa..c84650e4a7aa 100644
--- a/sound/soc/ti/davinci-vcif.c
+++ b/sound/soc/ti/davinci-vcif.c
@@ -1,23 +1,10 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
 /*
  * ALSA SoC Voice Codec Interface for TI DAVINCI processor
  *
  * Copyright (C) 2010 Texas Instruments.
  *
  * Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
  */
 
 #include <linux/init.h>
diff --git a/sound/soc/ti/edma-pcm.c b/sound/soc/ti/edma-pcm.c
index fdffb801b185..3ebea1bd15cb 100644
--- a/sound/soc/ti/edma-pcm.c
+++ b/sound/soc/ti/edma-pcm.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * edma-pcm.c - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx
  *
@@ -6,15 +7,6 @@
  * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * Based on: sound/soc/tegra/tegra_pcm.c
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
  */
 
 #include <linux/module.h>
diff --git a/sound/soc/ti/edma-pcm.h b/sound/soc/ti/edma-pcm.h
index 8058bdb0f032..941d18527d91 100644
--- a/sound/soc/ti/edma-pcm.h
+++ b/sound/soc/ti/edma-pcm.h
@@ -1,3 +1,4 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
 /*
  * edma-pcm.h - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx
  *
@@ -6,15 +7,6 @@
  * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * Based on: sound/soc/tegra/tegra_pcm.h
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
  */
 
 #ifndef __EDMA_PCM_H__
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 9cfefe44a75f..2c3f2a4c1700 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -1,24 +1,10 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * n810.c  --  SoC audio for Nokia N810
  *
  * Copyright (C) 2008 Nokia Corporation
  *
  * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/clk.h>
@@ -261,16 +247,19 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = {
 };
 
 /* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(aic33,
+	DAILINK_COMP_ARRAY(COMP_CPU("48076000.mcbsp")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("48076000.mcbsp")));
+
 static struct snd_soc_dai_link n810_dai = {
 	.name = "TLV320AIC33",
 	.stream_name = "AIC33",
-	.cpu_dai_name = "48076000.mcbsp",
-	.platform_name = "48076000.mcbsp",
-	.codec_name = "tlv320aic3x-codec.1-0018",
-	.codec_dai_name = "tlv320aic3x-hifi",
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 		   SND_SOC_DAIFMT_CBM_CFM,
 	.ops = &n810_ops,
+	SND_SOC_DAILINK_REG(aic33),
 };
 
 /* Audio machine driver */
diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c
index fed45b41f9d3..6d564ab5e437 100644
--- a/sound/soc/ti/omap-abe-twl6040.c
+++ b/sound/soc/ti/omap-abe-twl6040.c
@@ -1,23 +1,9 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * omap-abe-twl6040.c  --  SoC audio for TI OMAP based boards with ABE and
  *			   twl6040 codec
  *
  * Author: Misael Lopez Cruz <misael.lopez@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/clk.h>
@@ -35,6 +21,18 @@
 #include "omap-mcpdm.h"
 #include "../codecs/twl6040.h"
 
+SND_SOC_DAILINK_DEFS(link0,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl6040-codec",
+				      "twl6040-legacy")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(link1,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec",
+				      "dmic-hifi")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
 struct abe_twl6040 {
 	struct snd_soc_card card;
 	struct snd_soc_dai_link dai_links[2];
@@ -255,10 +253,14 @@ static int omap_abe_probe(struct platform_device *pdev)
 
 	priv->dai_links[0].name = "DMIC";
 	priv->dai_links[0].stream_name = "TWL6040";
-	priv->dai_links[0].cpu_of_node = dai_node;
-	priv->dai_links[0].platform_of_node = dai_node;
-	priv->dai_links[0].codec_dai_name = "twl6040-legacy";
-	priv->dai_links[0].codec_name = "twl6040-codec";
+	priv->dai_links[0].cpus = link0_cpus;
+	priv->dai_links[0].num_cpus = 1;
+	priv->dai_links[0].cpus->of_node = dai_node;
+	priv->dai_links[0].platforms = link0_platforms;
+	priv->dai_links[0].num_platforms = 1;
+	priv->dai_links[0].platforms->of_node = dai_node;
+	priv->dai_links[0].codecs = link0_codecs;
+	priv->dai_links[0].num_codecs = 1;
 	priv->dai_links[0].init = omap_abe_twl6040_init;
 	priv->dai_links[0].ops = &omap_abe_ops;
 
@@ -267,10 +269,14 @@ static int omap_abe_probe(struct platform_device *pdev)
 		num_links = 2;
 		priv->dai_links[1].name = "TWL6040";
 		priv->dai_links[1].stream_name = "DMIC Capture";
-		priv->dai_links[1].cpu_of_node = dai_node;
-		priv->dai_links[1].platform_of_node = dai_node;
-		priv->dai_links[1].codec_dai_name = "dmic-hifi";
-		priv->dai_links[1].codec_name = "dmic-codec";
+		priv->dai_links[1].cpus = link1_cpus;
+		priv->dai_links[1].num_cpus = 1;
+		priv->dai_links[1].cpus->of_node = dai_node;
+		priv->dai_links[1].platforms = link1_platforms;
+		priv->dai_links[1].num_platforms = 1;
+		priv->dai_links[1].platforms->of_node = dai_node;
+		priv->dai_links[1].codecs = link1_codecs;
+		priv->dai_links[1].num_codecs = 1;
 		priv->dai_links[1].init = omap_abe_dmic_init;
 		priv->dai_links[1].ops = &omap_abe_dmic_ops;
 	} else {
diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c
index cba9645b6487..3f226be123d4 100644
--- a/sound/soc/ti/omap-dmic.c
+++ b/sound/soc/ti/omap-dmic.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * omap-dmic.c  --  OMAP ASoC DMIC DAI driver
  *
@@ -7,21 +8,6 @@
  *	   Misael Lopez Cruz <misael.lopez@ti.com>
  *	   Liam Girdwood <lrg@ti.com>
  *	   Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/init.h>
diff --git a/sound/soc/ti/omap-dmic.h b/sound/soc/ti/omap-dmic.h
index 231e728bff0e..472cdbd9a0da 100644
--- a/sound/soc/ti/omap-dmic.h
+++ b/sound/soc/ti/omap-dmic.h
@@ -1,9 +1,6 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
 /*
  * omap-dmic.h  --  OMAP Digital Microphone Controller
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef _OMAP_DMIC_H
diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c
index 673a9eb153b2..def2a0ce8886 100644
--- a/sound/soc/ti/omap-hdmi.c
+++ b/sound/soc/ti/omap-hdmi.c
@@ -1,19 +1,10 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * omap-hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
  *
  * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
  *
  * Author: Jyri Sarha <jsarha@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
  */
 
 #include <linux/kernel.h>
@@ -321,6 +312,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 	struct hdmi_audio_data *ad;
 	struct snd_soc_dai_driver *dai_drv;
 	struct snd_soc_card *card;
+	struct snd_soc_dai_link_component *compnent;
 	int ret;
 
 	if (!ha) {
@@ -371,12 +363,23 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 		devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL);
 	if (!card->dai_link)
 		return -ENOMEM;
+
+	compnent = devm_kzalloc(dev, 3 * sizeof(*compnent), GFP_KERNEL);
+	if (!compnent)
+		return -ENOMEM;
+	card->dai_link->cpus		= &compnent[0];
+	card->dai_link->num_cpus	= 1;
+	card->dai_link->codecs		= &compnent[1];
+	card->dai_link->num_codecs	= 1;
+	card->dai_link->platforms	= &compnent[2];
+	card->dai_link->num_platforms	= 1;
+
 	card->dai_link->name = card->name;
 	card->dai_link->stream_name = card->name;
-	card->dai_link->cpu_dai_name = dev_name(ad->dssdev);
-	card->dai_link->platform_name = dev_name(ad->dssdev);
-	card->dai_link->codec_name = "snd-soc-dummy";
-	card->dai_link->codec_dai_name = "snd-soc-dummy-dai";
+	card->dai_link->cpus->dai_name = dev_name(ad->dssdev);
+	card->dai_link->platforms->name = dev_name(ad->dssdev);
+	card->dai_link->codecs->name = "snd-soc-dummy";
+	card->dai_link->codecs->dai_name = "snd-soc-dummy-dai";
 	card->num_links = 1;
 	card->dev = dev;
 
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index a395598f1f20..26b503bbdb5f 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * omap-mcbsp.c  --  OMAP ALSA SoC DAI driver using McBSP port
  *
@@ -5,21 +6,6 @@
  *
  * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
  *          Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/init.h>
@@ -1438,7 +1424,7 @@ static int asoc_mcbsp_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	return sdma_pcm_platform_register(&pdev->dev, NULL, NULL);
+	return sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
 }
 
 static int asoc_mcbsp_remove(struct platform_device *pdev)
diff --git a/sound/soc/ti/omap-mcbsp.h b/sound/soc/ti/omap-mcbsp.h
index 7911d24898c9..9fdba86ba255 100644
--- a/sound/soc/ti/omap-mcbsp.h
+++ b/sound/soc/ti/omap-mcbsp.h
@@ -1,3 +1,4 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
 /*
  * omap-mcbsp.h
  *
@@ -5,21 +6,6 @@
  *
  * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
  *          Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #ifndef __OMAP_MCBSP_H__
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index 7d5bdc5a2890..b8c8290265c7 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * omap-mcpdm.c  --  OMAP ALSA SoC DAI driver using McPDM port
  *
@@ -7,21 +8,6 @@
  * Contact: Jorge Eduardo Candelaria <x0107209@ti.com>
  *          Margarita Olaya <magi.olaya@ti.com>
  *          Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/init.h>
diff --git a/sound/soc/ti/omap-mcpdm.h b/sound/soc/ti/omap-mcpdm.h
index de8cf26595b1..e9956b4ef5e6 100644
--- a/sound/soc/ti/omap-mcpdm.h
+++ b/sound/soc/ti/omap-mcpdm.h
@@ -1,24 +1,10 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
 /*
  * omap-mcpdm.h
  *
  * Copyright (C) 2009 - 2011 Texas Instruments
  *
  * Contact: Misael Lopez Cruz <misael.lopez@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #ifndef __OMAP_MCPDM_H__
diff --git a/sound/soc/ti/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c
index cccc316743fa..92dbe2c67290 100644
--- a/sound/soc/ti/omap-twl4030.c
+++ b/sound/soc/ti/omap-twl4030.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * omap-twl4030.c  --  SoC audio for TI SoC based boards with twl4030 codec
  *
@@ -13,21 +14,6 @@
  * igep0020 (Author: Enric Balletbo i Serra <eballetbo@iseebcn.com>)
  * zoom2 (Author: Misael Lopez Cruz <misael.lopez@ti.com>)
  * sdp3430 (Author: Misael Lopez Cruz <misael.lopez@ti.com>)
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/platform_device.h>
@@ -209,26 +195,30 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd)
 }
 
 /* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(hifi,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.2")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.2")));
+
+SND_SOC_DAILINK_DEFS(voice,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.3")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-voice")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.3")));
+
 static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
 	{
 		.name = "TWL4030 HiFi",
 		.stream_name = "TWL4030 HiFi",
-		.cpu_dai_name = "omap-mcbsp.2",
-		.codec_dai_name = "twl4030-hifi",
-		.platform_name = "omap-mcbsp.2",
-		.codec_name = "twl4030-codec",
 		.init = omap_twl4030_init,
 		.ops = &omap_twl4030_ops,
+		SND_SOC_DAILINK_REG(hifi),
 	},
 	{
 		.name = "TWL4030 Voice",
 		.stream_name = "TWL4030 Voice",
-		.cpu_dai_name = "omap-mcbsp.3",
-		.codec_dai_name = "twl4030-voice",
-		.platform_name = "omap-mcbsp.3",
-		.codec_name = "twl4030-codec",
 		.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
 			   SND_SOC_DAIFMT_CBM_CFM,
+		SND_SOC_DAILINK_REG(voice),
 	},
 };
 
@@ -272,21 +262,21 @@ static int omap_twl4030_probe(struct platform_device *pdev)
 			dev_err(&pdev->dev, "McBSP node is not provided\n");
 			return -EINVAL;
 		}
-		omap_twl4030_dai_links[0].cpu_dai_name  = NULL;
-		omap_twl4030_dai_links[0].cpu_of_node = dai_node;
+		omap_twl4030_dai_links[0].cpus->dai_name  = NULL;
+		omap_twl4030_dai_links[0].cpus->of_node = dai_node;
 
-		omap_twl4030_dai_links[0].platform_name  = NULL;
-		omap_twl4030_dai_links[0].platform_of_node = dai_node;
+		omap_twl4030_dai_links[0].platforms->name  = NULL;
+		omap_twl4030_dai_links[0].platforms->of_node = dai_node;
 
 		dai_node = of_parse_phandle(node, "ti,mcbsp-voice", 0);
 		if (!dai_node) {
 			card->num_links = 1;
 		} else {
-			omap_twl4030_dai_links[1].cpu_dai_name  = NULL;
-			omap_twl4030_dai_links[1].cpu_of_node = dai_node;
+			omap_twl4030_dai_links[1].cpus->dai_name  = NULL;
+			omap_twl4030_dai_links[1].cpus->of_node = dai_node;
 
-			omap_twl4030_dai_links[1].platform_name  = NULL;
-			omap_twl4030_dai_links[1].platform_of_node = dai_node;
+			omap_twl4030_dai_links[1].platforms->name  = NULL;
+			omap_twl4030_dai_links[1].platforms->of_node = dai_node;
 		}
 
 		priv->jack_detect = of_get_named_gpio(node,
diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c
index 4e3de712159c..545f8dac9bd5 100644
--- a/sound/soc/ti/omap3pandora.c
+++ b/sound/soc/ti/omap3pandora.c
@@ -1,22 +1,8 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * omap3pandora.c  --  SoC audio for Pandora Handheld Console
  *
  * Author: Gražvydas Ignotas <notasas@gmail.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/clk.h>
@@ -189,29 +175,33 @@ static const struct snd_soc_ops omap3pandora_ops = {
 };
 
 /* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(out,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.2")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.2")));
+
+SND_SOC_DAILINK_DEFS(in,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.4")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.4")));
+
 static struct snd_soc_dai_link omap3pandora_dai[] = {
 	{
 		.name = "PCM1773",
 		.stream_name = "HiFi Out",
-		.cpu_dai_name = "omap-mcbsp.2",
-		.codec_dai_name = "twl4030-hifi",
-		.platform_name = "omap-mcbsp.2",
-		.codec_name = "twl4030-codec",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBS_CFS,
 		.ops = &omap3pandora_ops,
 		.init = omap3pandora_out_init,
+		SND_SOC_DAILINK_REG(out),
 	}, {
 		.name = "TWL4030",
 		.stream_name = "Line/Mic In",
-		.cpu_dai_name = "omap-mcbsp.4",
-		.codec_dai_name = "twl4030-hifi",
-		.platform_name = "omap-mcbsp.4",
-		.codec_name = "twl4030-codec",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBS_CFS,
 		.ops = &omap3pandora_ops,
 		.init = omap3pandora_in_init,
+		SND_SOC_DAILINK_REG(in),
 	}
 };
 
diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c
index e4096779ca05..1ca466bc4025 100644
--- a/sound/soc/ti/osk5912.c
+++ b/sound/soc/ti/osk5912.c
@@ -1,24 +1,10 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * osk5912.c  --  SoC audio for OSK 5912
  *
  * Copyright (C) 2008 Mistral Solutions
  *
  * Contact: Arun KS  <arunks@mistralsolutions.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/clk.h>
@@ -91,16 +77,19 @@ static const struct snd_soc_dapm_route audio_map[] = {
 };
 
 /* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(aic23,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic23-codec",
+				      "tlv320aic23-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.1")));
+
 static struct snd_soc_dai_link osk_dai = {
 	.name = "TLV320AIC23",
 	.stream_name = "AIC23",
-	.cpu_dai_name = "omap-mcbsp.1",
-	.codec_dai_name = "tlv320aic23-hifi",
-	.platform_name = "omap-mcbsp.1",
-	.codec_name = "tlv320aic23-codec",
 	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
 		   SND_SOC_DAIFMT_CBM_CFM,
 	.ops = &osk_ops,
+	SND_SOC_DAILINK_REG(aic23),
 };
 
 /* Audio machine driver */
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 57448bd5ad77..bc6046534fa5 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-only
 /*
  * rx51.c  --  SoC audio for Nokia RX-51
  *
@@ -6,21 +7,6 @@
  * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *          Eduardo Valentin <eduardo.valentin@nokia.com>
  *          Jarkko Nikula <jarkko.nikula@bitmer.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
  */
 
 #include <linux/delay.h>
@@ -312,18 +298,21 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 }
 
 /* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(aic34,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.2")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.2-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.2")));
+
 static struct snd_soc_dai_link rx51_dai[] = {
 	{
 		.name = "TLV320AIC34",
 		.stream_name = "AIC34",
-		.cpu_dai_name = "omap-mcbsp.2",
-		.codec_dai_name = "tlv320aic3x-hifi",
-		.platform_name = "omap-mcbsp.2",
-		.codec_name = "tlv320aic3x-codec.2-0018",
 		.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
 			   SND_SOC_DAIFMT_CBM_CFM,
 		.init = rx51_aic34_init,
 		.ops = &rx51_ops,
+		SND_SOC_DAILINK_REG(aic34),
 	},
 };
 
@@ -389,18 +378,18 @@ static int rx51_soc_probe(struct platform_device *pdev)
 			dev_err(&pdev->dev, "McBSP node is not provided\n");
 			return -EINVAL;
 		}
-		rx51_dai[0].cpu_dai_name = NULL;
-		rx51_dai[0].platform_name = NULL;
-		rx51_dai[0].cpu_of_node = dai_node;
-		rx51_dai[0].platform_of_node = dai_node;
+		rx51_dai[0].cpus->dai_name = NULL;
+		rx51_dai[0].platforms->name = NULL;
+		rx51_dai[0].cpus->of_node = dai_node;
+		rx51_dai[0].platforms->of_node = dai_node;
 
 		dai_node = of_parse_phandle(np, "nokia,audio-codec", 0);
 		if (!dai_node) {
 			dev_err(&pdev->dev, "Codec node is not provided\n");
 			return -EINVAL;
 		}
-		rx51_dai[0].codec_name = NULL;
-		rx51_dai[0].codec_of_node = dai_node;
+		rx51_dai[0].codecs->name = NULL;
+		rx51_dai[0].codecs->of_node = dai_node;
 
 		dai_node = of_parse_phandle(np, "nokia,audio-codec", 1);
 		if (!dai_node) {