From 99c92ae4ffca81f4dfba3b7648734c56d0b32d4c Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 Nov 2011 18:21:14 -0700
Subject: ASoC: Tegra PCM: Use module_platform_driver

This saves some boiler-plate code.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_pcm.c | 13 +------------
 1 file changed, 1 insertion(+), 12 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 436def1dfa39..90345ee138f3 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -392,18 +392,7 @@ static struct platform_driver tegra_pcm_driver = {
 	.probe = tegra_pcm_platform_probe,
 	.remove = __devexit_p(tegra_pcm_platform_remove),
 };
-
-static int __init snd_tegra_pcm_init(void)
-{
-	return platform_driver_register(&tegra_pcm_driver);
-}
-module_init(snd_tegra_pcm_init);
-
-static void __exit snd_tegra_pcm_exit(void)
-{
-	platform_driver_unregister(&tegra_pcm_driver);
-}
-module_exit(snd_tegra_pcm_exit);
+module_platform_driver(tegra_pcm_driver);
 
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra PCM ASoC driver");
-- 
cgit v1.2.3


From f2296d7bf19a210a462a57bb90b1c9263d18a4ee Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 Nov 2011 18:21:15 -0700
Subject: ASoC: Tegra DAS: Use devm_ APIs and module_platform_driver

module_platform_drive saves some boiler-plate code.

The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_das.c | 45 ++++++++++-----------------------------------
 1 file changed, 10 insertions(+), 35 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c
index 3b55a44146af..fa3a4426cbdd 100644
--- a/sound/soc/tegra/tegra_das.c
+++ b/sound/soc/tegra/tegra_das.c
@@ -172,11 +172,11 @@ static int __devinit tegra_das_probe(struct platform_device *pdev)
 	if (das)
 		return -ENODEV;
 
-	das = kzalloc(sizeof(struct tegra_das), GFP_KERNEL);
+	das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL);
 	if (!das) {
 		dev_err(&pdev->dev, "Can't allocate tegra_das\n");
 		ret = -ENOMEM;
-		goto exit;
+		goto err;
 	}
 	das->dev = &pdev->dev;
 
@@ -184,22 +184,22 @@ static int __devinit tegra_das_probe(struct platform_device *pdev)
 	if (!res) {
 		dev_err(&pdev->dev, "No memory resource\n");
 		ret = -ENODEV;
-		goto err_free;
+		goto err;
 	}
 
-	region = request_mem_region(res->start, resource_size(res),
-					pdev->name);
+	region = devm_request_mem_region(&pdev->dev, res->start,
+					 resource_size(res), pdev->name);
 	if (!region) {
 		dev_err(&pdev->dev, "Memory region already claimed\n");
 		ret = -EBUSY;
-		goto err_free;
+		goto err;
 	}
 
-	das->regs = ioremap(res->start, resource_size(res));
+	das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res));
 	if (!das->regs) {
 		dev_err(&pdev->dev, "ioremap failed\n");
 		ret = -ENOMEM;
-		goto err_release;
+		goto err;
 	}
 
 	tegra_das_debug_add(das);
@@ -208,32 +208,18 @@ static int __devinit tegra_das_probe(struct platform_device *pdev)
 
 	return 0;
 
-err_release:
-	release_mem_region(res->start, resource_size(res));
-err_free:
-	kfree(das);
+err:
 	das = NULL;
-exit:
 	return ret;
 }
 
 static int __devexit tegra_das_remove(struct platform_device *pdev)
 {
-	struct resource *res;
-
 	if (!das)
 		return -ENODEV;
 
-	platform_set_drvdata(pdev, NULL);
-
 	tegra_das_debug_remove(das);
 
-	iounmap(das->regs);
-
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(res->start, resource_size(res));
-
-	kfree(das);
 	das = NULL;
 
 	return 0;
@@ -246,18 +232,7 @@ static struct platform_driver tegra_das_driver = {
 		.name = DRV_NAME,
 	},
 };
-
-static int __init tegra_das_modinit(void)
-{
-	return platform_driver_register(&tegra_das_driver);
-}
-module_init(tegra_das_modinit);
-
-static void __exit tegra_das_modexit(void)
-{
-	platform_driver_unregister(&tegra_das_driver);
-}
-module_exit(tegra_das_modexit);
+module_platform_driver(tegra_das_driver);
 
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra DAS driver");
-- 
cgit v1.2.3


From 65713ce8442b42c6f688bd8b0950a49d8f4dcf5f Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 Nov 2011 18:21:13 -0700
Subject: ASoC: Tegra: Move DAS configuration into machine drivers

This removes potentially machine-specific routing knowledge from the
I2S driverinto the machine drivers, which is better equipped to know
what the appropriate routing configuration is.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_i2s.c    | 18 ------------------
 sound/soc/tegra/tegra_wm8903.c | 13 +++++++++++++
 sound/soc/tegra/trimslice.c    | 15 +++++++++++++++
 3 files changed, 28 insertions(+), 18 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6728fab8c411..33e62fcdfce3 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -42,7 +42,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include "tegra_das.h"
 #include "tegra_i2s.h"
 
 #define DRV_NAME "tegra-i2s"
@@ -363,23 +362,6 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	/*
-	 * FIXME: Until a codec driver exists for the tegra DAS, hard-code a
-	 * 1:1 mapping between audio controllers and audio ports.
-	 */
-	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1 + pdev->id,
-					TEGRA_DAS_DAP_SEL_DAC1 + pdev->id);
-	if (ret) {
-		dev_err(&pdev->dev, "Can't set up DAP connection\n");
-		return ret;
-	}
-	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1 + pdev->id,
-					TEGRA_DAS_DAC_SEL_DAP1 + pdev->id);
-	if (ret) {
-		dev_err(&pdev->dev, "Can't set up DAC connection\n");
-		return ret;
-	}
-
 	i2s = kzalloc(sizeof(struct tegra_i2s), GFP_KERNEL);
 	if (!i2s) {
 		dev_err(&pdev->dev, "Can't allocate tegra_i2s\n");
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index a81cf39257bf..9b0ee1510935 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -249,6 +249,19 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	struct tegra_wm8903_platform_data *pdata = machine->pdata;
 	int ret;
 
+	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1,
+					   TEGRA_DAS_DAP_SEL_DAC1);
+	if (ret) {
+		dev_err(card->dev, "Can't set up DAS DAP connection\n");
+		return ret;
+	}
+	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1,
+					   TEGRA_DAS_DAC_SEL_DAP1);
+	if (ret) {
+		dev_err(card->dev, "Can't set up DAS DAC connection\n");
+		return ret;
+	}
+
 	if (gpio_is_valid(pdata->gpio_spkr_en)) {
 		ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
 		if (ret) {
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index b3a7efa6d960..2699a6fa45f9 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -118,7 +118,22 @@ static const struct snd_soc_dapm_route trimslice_audio_map[] = {
 static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = codec->card;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1,
+					   TEGRA_DAS_DAP_SEL_DAC1);
+	if (ret) {
+		dev_err(card->dev, "Can't set up DAS DAP connection\n");
+		return ret;
+	}
+	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1,
+					   TEGRA_DAS_DAC_SEL_DAP1);
+	if (ret) {
+		dev_err(card->dev, "Can't set up DAS DAC connection\n");
+		return ret;
+	}
 
 	snd_soc_dapm_nc_pin(dapm, "LHPOUT");
 	snd_soc_dapm_nc_pin(dapm, "RHPOUT");
-- 
cgit v1.2.3


From bea0ed0825be288f9fc98696fc476066776b26be Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 Nov 2011 18:21:16 -0700
Subject: ASoC: Tegra I2S: Use devm_ APIs and module_platform_driver

module_platform_drive saves some boiler-plate code.

The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_i2s.c | 45 ++++++++++-----------------------------------
 1 file changed, 10 insertions(+), 35 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 33e62fcdfce3..76014f0d8a29 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -362,11 +362,11 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	i2s = kzalloc(sizeof(struct tegra_i2s), GFP_KERNEL);
+	i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL);
 	if (!i2s) {
 		dev_err(&pdev->dev, "Can't allocate tegra_i2s\n");
 		ret = -ENOMEM;
-		goto exit;
+		goto err;
 	}
 	dev_set_drvdata(&pdev->dev, i2s);
 
@@ -374,7 +374,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 	if (IS_ERR(i2s->clk_i2s)) {
 		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
 		ret = PTR_ERR(i2s->clk_i2s);
-		goto err_free;
+		goto err;
 	}
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -391,19 +391,19 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 		goto err_clk_put;
 	}
 
-	memregion = request_mem_region(mem->start, resource_size(mem),
-					DRV_NAME);
+	memregion = devm_request_mem_region(&pdev->dev, mem->start,
+					    resource_size(mem), DRV_NAME);
 	if (!memregion) {
 		dev_err(&pdev->dev, "Memory region already claimed\n");
 		ret = -EBUSY;
 		goto err_clk_put;
 	}
 
-	i2s->regs = ioremap(mem->start, resource_size(mem));
+	i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
 	if (!i2s->regs) {
 		dev_err(&pdev->dev, "ioremap failed\n");
 		ret = -ENOMEM;
-		goto err_release;
+		goto err_clk_put;
 	}
 
 	i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2;
@@ -422,43 +422,29 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
 		ret = -ENOMEM;
-		goto err_unmap;
+		goto err_clk_put;
 	}
 
 	tegra_i2s_debug_add(i2s, pdev->id);
 
 	return 0;
 
-err_unmap:
-	iounmap(i2s->regs);
-err_release:
-	release_mem_region(mem->start, resource_size(mem));
 err_clk_put:
 	clk_put(i2s->clk_i2s);
-err_free:
-	kfree(i2s);
-exit:
+err:
 	return ret;
 }
 
 static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev)
 {
 	struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev);
-	struct resource *res;
 
 	snd_soc_unregister_dai(&pdev->dev);
 
 	tegra_i2s_debug_remove(i2s);
 
-	iounmap(i2s->regs);
-
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(res->start, resource_size(res));
-
 	clk_put(i2s->clk_i2s);
 
-	kfree(i2s);
-
 	return 0;
 }
 
@@ -470,18 +456,7 @@ static struct platform_driver tegra_i2s_driver = {
 	.probe = tegra_i2s_platform_probe,
 	.remove = __devexit_p(tegra_i2s_platform_remove),
 };
-
-static int __init snd_tegra_i2s_init(void)
-{
-	return platform_driver_register(&tegra_i2s_driver);
-}
-module_init(snd_tegra_i2s_init);
-
-static void __exit snd_tegra_i2s_exit(void)
-{
-	platform_driver_unregister(&tegra_i2s_driver);
-}
-module_exit(snd_tegra_i2s_exit);
+module_platform_driver(tegra_i2s_driver);
 
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra I2S ASoC driver");
-- 
cgit v1.2.3


From 85e7652d89293a6dab42bfd31f276f8bc072d4c5 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 23 Nov 2011 11:40:40 +0100
Subject: ASoC: Constify snd_soc_dai_ops structs

Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.

The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel_ssc_dai.c   | 2 +-
 sound/soc/au1x/ac97c.c            | 2 +-
 sound/soc/au1x/i2sc.c             | 2 +-
 sound/soc/au1x/psc-ac97.c         | 2 +-
 sound/soc/au1x/psc-i2s.c          | 2 +-
 sound/soc/blackfin/bf5xx-i2s.c    | 2 +-
 sound/soc/blackfin/bf5xx-tdm.c    | 2 +-
 sound/soc/codecs/88pm860x-codec.c | 4 ++--
 sound/soc/codecs/ac97.c           | 2 +-
 sound/soc/codecs/ad1836.c         | 2 +-
 sound/soc/codecs/ad193x.c         | 2 +-
 sound/soc/codecs/adau1373.c       | 2 +-
 sound/soc/codecs/adau1701.c       | 2 +-
 sound/soc/codecs/adav80x.c        | 2 +-
 sound/soc/codecs/ak4104.c         | 2 +-
 sound/soc/codecs/ak4535.c         | 2 +-
 sound/soc/codecs/ak4641.c         | 4 ++--
 sound/soc/codecs/ak4642.c         | 2 +-
 sound/soc/codecs/ak4671.c         | 2 +-
 sound/soc/codecs/alc5623.c        | 2 +-
 sound/soc/codecs/alc5632.c        | 2 +-
 sound/soc/codecs/cq93vc.c         | 2 +-
 sound/soc/codecs/cs4270.c         | 2 +-
 sound/soc/codecs/cs4271.c         | 2 +-
 sound/soc/codecs/cs42l51.c        | 2 +-
 sound/soc/codecs/cs42l73.c        | 2 +-
 sound/soc/codecs/da7210.c         | 2 +-
 sound/soc/codecs/jz4740.c         | 2 +-
 sound/soc/codecs/max98088.c       | 4 ++--
 sound/soc/codecs/max98095.c       | 6 +++---
 sound/soc/codecs/max9850.c        | 2 +-
 sound/soc/codecs/rt5631.c         | 2 +-
 sound/soc/codecs/sgtl5000.c       | 2 +-
 sound/soc/codecs/sn95031.c        | 8 ++++----
 sound/soc/codecs/ssm2602.c        | 2 +-
 sound/soc/codecs/sta32x.c         | 2 +-
 sound/soc/codecs/stac9766.c       | 4 ++--
 sound/soc/codecs/tlv320aic23.c    | 2 +-
 sound/soc/codecs/tlv320aic26.c    | 2 +-
 sound/soc/codecs/tlv320aic32x4.c  | 2 +-
 sound/soc/codecs/tlv320aic3x.c    | 2 +-
 sound/soc/codecs/tlv320dac33.c    | 2 +-
 sound/soc/codecs/twl4030.c        | 4 ++--
 sound/soc/codecs/twl6040.c        | 2 +-
 sound/soc/codecs/uda134x.c        | 2 +-
 sound/soc/codecs/uda1380.c        | 6 +++---
 sound/soc/codecs/wl1273.c         | 2 +-
 sound/soc/codecs/wm5100.c         | 2 +-
 sound/soc/codecs/wm8350.c         | 2 +-
 sound/soc/codecs/wm8400.c         | 2 +-
 sound/soc/codecs/wm8510.c         | 2 +-
 sound/soc/codecs/wm8523.c         | 2 +-
 sound/soc/codecs/wm8580.c         | 4 ++--
 sound/soc/codecs/wm8711.c         | 2 +-
 sound/soc/codecs/wm8728.c         | 2 +-
 sound/soc/codecs/wm8731.c         | 2 +-
 sound/soc/codecs/wm8737.c         | 2 +-
 sound/soc/codecs/wm8741.c         | 2 +-
 sound/soc/codecs/wm8750.c         | 2 +-
 sound/soc/codecs/wm8753.c         | 4 ++--
 sound/soc/codecs/wm8770.c         | 2 +-
 sound/soc/codecs/wm8776.c         | 4 ++--
 sound/soc/codecs/wm8804.c         | 2 +-
 sound/soc/codecs/wm8900.c         | 2 +-
 sound/soc/codecs/wm8903.c         | 2 +-
 sound/soc/codecs/wm8904.c         | 2 +-
 sound/soc/codecs/wm8940.c         | 2 +-
 sound/soc/codecs/wm8955.c         | 2 +-
 sound/soc/codecs/wm8960.c         | 2 +-
 sound/soc/codecs/wm8961.c         | 2 +-
 sound/soc/codecs/wm8962.c         | 2 +-
 sound/soc/codecs/wm8971.c         | 2 +-
 sound/soc/codecs/wm8974.c         | 2 +-
 sound/soc/codecs/wm8978.c         | 2 +-
 sound/soc/codecs/wm8983.c         | 2 +-
 sound/soc/codecs/wm8985.c         | 2 +-
 sound/soc/codecs/wm8988.c         | 2 +-
 sound/soc/codecs/wm8990.c         | 2 +-
 sound/soc/codecs/wm8991.c         | 2 +-
 sound/soc/codecs/wm8993.c         | 2 +-
 sound/soc/codecs/wm8994.c         | 6 +++---
 sound/soc/codecs/wm8995.c         | 6 +++---
 sound/soc/codecs/wm8996.c         | 2 +-
 sound/soc/codecs/wm9081.c         | 2 +-
 sound/soc/codecs/wm9705.c         | 2 +-
 sound/soc/codecs/wm9712.c         | 4 ++--
 sound/soc/codecs/wm9713.c         | 6 +++---
 sound/soc/davinci/davinci-i2s.c   | 2 +-
 sound/soc/davinci/davinci-mcasp.c | 2 +-
 sound/soc/davinci/davinci-vcif.c  | 2 +-
 sound/soc/ep93xx/ep93xx-ac97.c    | 2 +-
 sound/soc/ep93xx/ep93xx-i2s.c     | 2 +-
 sound/soc/fsl/fsl_ssi.c           | 2 +-
 sound/soc/fsl/mpc5200_psc_ac97.c  | 4 ++--
 sound/soc/fsl/mpc5200_psc_i2s.c   | 2 +-
 sound/soc/imx/imx-ssi.c           | 2 +-
 sound/soc/jz4740/jz4740-i2s.c     | 2 +-
 sound/soc/kirkwood/kirkwood-i2s.c | 2 +-
 sound/soc/mxs/mxs-saif.c          | 2 +-
 sound/soc/nuc900/nuc900-ac97.c    | 2 +-
 sound/soc/omap/ams-delta.c        | 2 +-
 sound/soc/omap/omap-hdmi.c        | 2 +-
 sound/soc/omap/omap-mcbsp.c       | 2 +-
 sound/soc/omap/omap-mcpdm.c       | 2 +-
 sound/soc/pxa/pxa-ssp.c           | 2 +-
 sound/soc/pxa/pxa2xx-ac97.c       | 6 +++---
 sound/soc/pxa/pxa2xx-i2s.c        | 2 +-
 sound/soc/s6000/s6000-i2s.c       | 2 +-
 sound/soc/samsung/ac97.c          | 4 ++--
 sound/soc/samsung/i2s.c           | 2 +-
 sound/soc/samsung/pcm.c           | 2 +-
 sound/soc/samsung/s3c2412-i2s.c   | 2 +-
 sound/soc/samsung/s3c24xx-i2s.c   | 2 +-
 sound/soc/samsung/spdif.c         | 2 +-
 sound/soc/sh/fsi.c                | 2 +-
 sound/soc/sh/hac.c                | 2 +-
 sound/soc/sh/siu_dai.c            | 2 +-
 sound/soc/sh/ssi.c                | 2 +-
 sound/soc/soc-core.c              | 2 +-
 sound/soc/tegra/tegra_i2s.c       | 2 +-
 sound/soc/tegra/tegra_spdif.c     | 2 +-
 121 files changed, 147 insertions(+), 147 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 71225090c49f..a67fc9b7dbe7 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -719,7 +719,7 @@ static int atmel_ssc_remove(struct snd_soc_dai *dai)
 #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
 			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops atmel_ssc_dai_ops = {
+static const struct snd_soc_dai_ops atmel_ssc_dai_ops = {
 	.startup	= atmel_ssc_startup,
 	.shutdown	= atmel_ssc_shutdown,
 	.prepare	= atmel_ssc_prepare,
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
index 726bd651a105..7771934b93e2 100644
--- a/sound/soc/au1x/ac97c.c
+++ b/sound/soc/au1x/ac97c.c
@@ -195,7 +195,7 @@ static int alchemy_ac97c_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops alchemy_ac97c_ops = {
+static const struct snd_soc_dai_ops alchemy_ac97c_ops = {
 	.startup		= alchemy_ac97c_startup,
 };
 
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
index 6bcf48f5884c..2d5f755ac99c 100644
--- a/sound/soc/au1x/i2sc.c
+++ b/sound/soc/au1x/i2sc.c
@@ -201,7 +201,7 @@ static int au1xi2s_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
+static const const struct snd_soc_dai_ops au1xi2s_dai_ops = {
 	.startup	= au1xi2s_startup,
 	.trigger	= au1xi2s_trigger,
 	.hw_params	= au1xi2s_hw_params,
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 0c6acd547141..87daf456b1c9 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -337,7 +337,7 @@ static int au1xpsc_ac97_probe(struct snd_soc_dai *dai)
 	return au1xpsc_ac97_workdata ? 0 : -ENODEV;
 }
 
-static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+static const struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
 	.startup	= au1xpsc_ac97_startup,
 	.trigger	= au1xpsc_ac97_trigger,
 	.hw_params	= au1xpsc_ac97_hw_params,
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index e03c5ce01b30..f7714d50bdaf 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -265,7 +265,7 @@ static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+static const struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
 	.startup	= au1xpsc_i2s_startup,
 	.trigger	= au1xpsc_i2s_trigger,
 	.hw_params	= au1xpsc_i2s_hw_params,
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 00cc3e00b2fe..b31662e3a428 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -223,7 +223,7 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai)
 	 SNDRV_PCM_FMTBIT_S24_LE | \
 	 SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
+static const struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
 	.shutdown	= bf5xx_i2s_shutdown,
 	.hw_params	= bf5xx_i2s_hw_params,
 	.set_fmt	= bf5xx_i2s_set_dai_fmt,
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index a822d1ee1380..7876b5090fda 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -226,7 +226,7 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai)
 #define bf5xx_tdm_resume       NULL
 #endif
 
-static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = {
+static const struct snd_soc_dai_ops bf5xx_tdm_dai_ops = {
 	.hw_params      = bf5xx_tdm_hw_params,
 	.set_fmt        = bf5xx_tdm_set_dai_fmt,
 	.shutdown       = bf5xx_tdm_shutdown,
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 5ca122e51183..ea305b88cb55 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1198,14 +1198,14 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
+static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
 	.digital_mute	= pm860x_digital_mute,
 	.hw_params	= pm860x_pcm_hw_params,
 	.set_fmt	= pm860x_pcm_set_dai_fmt,
 	.set_sysclk	= pm860x_set_dai_sysclk,
 };
 
-static struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
+static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
 	.digital_mute	= pm860x_digital_mute,
 	.hw_params	= pm860x_i2s_hw_params,
 	.set_fmt	= pm860x_i2s_set_dai_fmt,
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index e715186b4300..8f3216793eb5 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -39,7 +39,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
-static struct snd_soc_dai_ops ac97_dai_ops = {
+static const struct snd_soc_dai_ops ac97_dai_ops = {
 	.prepare	= ac97_prepare,
 };
 
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 4e5c5726366b..fab0948f7a54 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -189,7 +189,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops ad1836_dai_ops = {
+static const struct snd_soc_dai_ops ad1836_dai_ops = {
 	.hw_params = ad1836_hw_params,
 	.set_fmt = ad1836_set_dai_fmt,
 };
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 120602130b5c..1901cd222233 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -312,7 +312,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops ad193x_dai_ops = {
+static const struct snd_soc_dai_ops ad193x_dai_ops = {
 	.hw_params = ad193x_hw_params,
 	.digital_mute = ad193x_mute,
 	.set_tdm_slot = ad193x_set_tdm_slot,
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 45c63028b40d..2e040af9ad57 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1042,7 +1042,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static const struct snd_soc_dai_ops adau1373_dai_ops = {
+static const const struct snd_soc_dai_ops adau1373_dai_ops = {
 	.hw_params	= adau1373_hw_params,
 	.set_sysclk	= adau1373_set_dai_sysclk,
 	.set_fmt	= adau1373_set_dai_fmt,
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 8b7e1c50d6e9..c69bdfe745bb 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -427,7 +427,7 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 #define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static const struct snd_soc_dai_ops adau1701_dai_ops = {
+static const const struct snd_soc_dai_ops adau1701_dai_ops = {
 	.set_fmt	= adau1701_set_dai_fmt,
 	.hw_params	= adau1701_hw_params,
 	.digital_mute	= adau1701_digital_mute,
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index f9f08948e5e8..d927febd02cc 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -718,7 +718,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
 		adav80x->rate = 0;
 }
 
-static const struct snd_soc_dai_ops adav80x_dai_ops = {
+static const const struct snd_soc_dai_ops adav80x_dai_ops = {
 	.set_fmt = adav80x_set_dai_fmt,
 	.hw_params = adav80x_hw_params,
 	.startup = adav80x_dai_startup,
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index d3b29dce6ed7..152420ca78b8 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -170,7 +170,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
 	return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val);
 }
 
-static struct snd_soc_dai_ops ak4101_dai_ops = {
+static const struct snd_soc_dai_ops ak4101_dai_ops = {
 	.hw_params = ak4104_hw_params,
 	.set_fmt = ak4104_set_dai_fmt,
 };
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 95d782d86e7d..f6c47345bcc8 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -331,7 +331,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
-static struct snd_soc_dai_ops ak4535_dai_ops = {
+static const struct snd_soc_dai_ops ak4535_dai_ops = {
 	.hw_params	= ak4535_hw_params,
 	.set_fmt	= ak4535_set_dai_fmt,
 	.digital_mute	= ak4535_mute,
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 77838586f358..3657c76cc127 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -442,14 +442,14 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 			 SNDRV_PCM_RATE_16000)
 #define AK4641_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE)
 
-static struct snd_soc_dai_ops ak4641_i2s_dai_ops = {
+static const struct snd_soc_dai_ops ak4641_i2s_dai_ops = {
 	.hw_params    = ak4641_i2s_hw_params,
 	.set_fmt      = ak4641_i2s_set_dai_fmt,
 	.digital_mute = ak4641_mute,
 	.set_sysclk   = ak4641_set_dai_sysclk,
 };
 
-static struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
+static const struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
 	.hw_params    = NULL, /* rates are controlled by BT chip */
 	.set_fmt      = ak4641_pcm_set_dai_fmt,
 	.digital_mute = ak4641_mute,
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 859e0155e18d..c887ddf1061e 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -435,7 +435,7 @@ static int ak4642_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static struct snd_soc_dai_ops ak4642_dai_ops = {
+static const struct snd_soc_dai_ops ak4642_dai_ops = {
 	.startup	= ak4642_dai_startup,
 	.shutdown	= ak4642_dai_shutdown,
 	.set_sysclk	= ak4642_dai_set_sysclk,
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index de9ff66d3721..4f5c69f735a9 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -594,7 +594,7 @@ static int ak4671_set_bias_level(struct snd_soc_codec *codec,
 
 #define AK4671_FORMATS		SNDRV_PCM_FMTBIT_S16_LE
 
-static struct snd_soc_dai_ops ak4671_dai_ops = {
+static const struct snd_soc_dai_ops ak4671_dai_ops = {
 	.hw_params	= ak4671_hw_params,
 	.set_sysclk	= ak4671_set_dai_sysclk,
 	.set_fmt	= ak4671_set_dai_fmt,
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 984b14bcb605..88647d3ab24b 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -839,7 +839,7 @@ static int alc5623_set_bias_level(struct snd_soc_codec *codec,
 			| SNDRV_PCM_FMTBIT_S24_LE \
 			| SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops alc5623_dai_ops = {
+static const struct snd_soc_dai_ops alc5623_dai_ops = {
 		.hw_params = alc5623_pcm_hw_params,
 		.digital_mute = alc5623_mute,
 		.set_fmt = alc5623_set_dai_fmt,
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 2d77665eb854..3f750def8967 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -924,7 +924,7 @@ static int alc5632_set_bias_level(struct snd_soc_codec *codec,
 			| SNDRV_PCM_FMTBIT_S24_LE \
 			| SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops alc5632_dai_ops = {
+static const struct snd_soc_dai_ops alc5632_dai_ops = {
 		.hw_params = alc5632_pcm_hw_params,
 		.digital_mute = alc5632_mute,
 		.set_fmt = alc5632_set_dai_fmt,
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd067f79..cbb3028e2008 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -122,7 +122,7 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
 #define CQ93VC_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
 #define CQ93VC_FORMATS	(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
 
-static struct snd_soc_dai_ops cq93vc_dai_ops = {
+static const struct snd_soc_dai_ops cq93vc_dai_ops = {
 	.digital_mute	= cq93vc_mute,
 	.set_sysclk	= cq93vc_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 73f46eb459f1..5396b91fa5f1 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -447,7 +447,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 		snd_soc_get_volsw, cs4270_soc_put_mute),
 };
 
-static struct snd_soc_dai_ops cs4270_dai_ops = {
+static const struct snd_soc_dai_ops cs4270_dai_ops = {
 	.hw_params	= cs4270_hw_params,
 	.set_sysclk	= cs4270_set_dai_sysclk,
 	.set_fmt	= cs4270_set_dai_fmt,
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 69fde1506fe1..a6f77a855f45 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -402,7 +402,7 @@ static const struct snd_kcontrol_new cs4271_snd_controls[] = {
 		7, 1, 1),
 };
 
-static struct snd_soc_dai_ops cs4271_dai_ops = {
+static const struct snd_soc_dai_ops cs4271_dai_ops = {
 	.hw_params	= cs4271_hw_params,
 	.set_sysclk	= cs4271_set_dai_sysclk,
 	.set_fmt	= cs4271_set_dai_fmt,
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 00718b5e747b..e378c4d52027 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -483,7 +483,7 @@ static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute)
 	return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg);
 }
 
-static struct snd_soc_dai_ops cs42l51_dai_ops = {
+static const struct snd_soc_dai_ops cs42l51_dai_ops = {
 	.hw_params      = cs42l51_hw_params,
 	.set_sysclk     = cs42l51_set_dai_sysclk,
 	.set_fmt        = cs42l51_set_dai_fmt,
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index d09578f397da..75d80b2e1ec4 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1190,7 +1190,7 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream,
 #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static const struct snd_soc_dai_ops cs42l73_ops = {
+static const const struct snd_soc_dai_ops cs42l73_ops = {
 	.startup = cs42l73_pcm_startup,
 	.hw_params = cs42l73_pcm_hw_params,
 	.set_fmt = cs42l73_set_dai_fmt,
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 8b5848a6374c..8ef820fd68c7 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -761,7 +761,7 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute)
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 /* DAI operations */
-static struct snd_soc_dai_ops da7210_dai_ops = {
+static const struct snd_soc_dai_ops da7210_dai_ops = {
 	.hw_params	= da7210_hw_params,
 	.set_fmt	= da7210_set_dai_fmt,
 	.digital_mute	= da7210_mute,
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index e373f8f06907..64a479c3429a 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -206,7 +206,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops jz4740_codec_dai_ops = {
+static const struct snd_soc_dai_ops jz4740_codec_dai_ops = {
 	.hw_params = jz4740_codec_hw_params,
 };
 
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index ebbf63c79c34..48a52a1aaaaa 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1650,14 +1650,14 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec,
 #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000
 #define MAX98088_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops max98088_dai1_ops = {
+static const struct snd_soc_dai_ops max98088_dai1_ops = {
        .set_sysclk = max98088_dai_set_sysclk,
        .set_fmt = max98088_dai1_set_fmt,
        .hw_params = max98088_dai1_hw_params,
        .digital_mute = max98088_dai1_digital_mute,
 };
 
-static struct snd_soc_dai_ops max98088_dai2_ops = {
+static const struct snd_soc_dai_ops max98088_dai2_ops = {
        .set_sysclk = max98088_dai_set_sysclk,
        .set_fmt = max98088_dai2_set_fmt,
        .hw_params = max98088_dai2_hw_params,
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 26d7b089fb9c..cc712d59ab64 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1782,19 +1782,19 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec,
 #define MAX98095_RATES SNDRV_PCM_RATE_8000_96000
 #define MAX98095_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops max98095_dai1_ops = {
+static const struct snd_soc_dai_ops max98095_dai1_ops = {
 	.set_sysclk = max98095_dai_set_sysclk,
 	.set_fmt = max98095_dai1_set_fmt,
 	.hw_params = max98095_dai1_hw_params,
 };
 
-static struct snd_soc_dai_ops max98095_dai2_ops = {
+static const struct snd_soc_dai_ops max98095_dai2_ops = {
 	.set_sysclk = max98095_dai_set_sysclk,
 	.set_fmt = max98095_dai2_set_fmt,
 	.hw_params = max98095_dai2_hw_params,
 };
 
-static struct snd_soc_dai_ops max98095_dai3_ops = {
+static const struct snd_soc_dai_ops max98095_dai3_ops = {
 	.set_sysclk = max98095_dai_set_sysclk,
 	.set_fmt = max98095_dai3_set_fmt,
 	.hw_params = max98095_dai3_hw_params,
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 208d2ee61855..94c2b586ed5d 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -254,7 +254,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec,
 #define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops max9850_dai_ops = {
+static const struct snd_soc_dai_ops max9850_dai_ops = {
 	.hw_params	= max9850_hw_params,
 	.set_sysclk	= max9850_set_dai_sysclk,
 	.set_fmt	= max9850_set_dai_fmt,
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 4646e808b90a..dac4d05f512d 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1664,7 +1664,7 @@ static int rt5631_resume(struct snd_soc_codec *codec)
 			SNDRV_PCM_FMTBIT_S24_LE | \
 			SNDRV_PCM_FMTBIT_S8)
 
-static struct snd_soc_dai_ops rt5631_ops = {
+static const struct snd_soc_dai_ops rt5631_ops = {
 	.hw_params = rt5631_hifi_pcm_params,
 	.set_fmt = rt5631_hifi_codec_set_dai_fmt,
 	.set_sysclk = rt5631_hifi_codec_set_dai_sysclk,
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index bbcf921166f7..1a6564b3684e 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -923,7 +923,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
 			SNDRV_PCM_FMTBIT_S24_LE |\
 			SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops sgtl5000_ops = {
+static const struct snd_soc_dai_ops sgtl5000_ops = {
 	.hw_params = sgtl5000_pcm_hw_params,
 	.digital_mute = sgtl5000_digital_mute,
 	.set_fmt = sgtl5000_set_dai_fmt,
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 887d618f4a63..65f2ef986c4f 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -698,21 +698,21 @@ static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
 }
 
 /* Codec DAI section */
-static struct snd_soc_dai_ops sn95031_headset_dai_ops = {
+static const struct snd_soc_dai_ops sn95031_headset_dai_ops = {
 	.digital_mute	= sn95031_pcm_hs_mute,
 	.hw_params	= sn95031_pcm_hw_params,
 };
 
-static struct snd_soc_dai_ops sn95031_speaker_dai_ops = {
+static const struct snd_soc_dai_ops sn95031_speaker_dai_ops = {
 	.digital_mute	= sn95031_pcm_spkr_mute,
 	.hw_params	= sn95031_pcm_hw_params,
 };
 
-static struct snd_soc_dai_ops sn95031_vib1_dai_ops = {
+static const struct snd_soc_dai_ops sn95031_vib1_dai_ops = {
 	.hw_params	= sn95031_pcm_hw_params,
 };
 
-static struct snd_soc_dai_ops sn95031_vib2_dai_ops = {
+static const struct snd_soc_dai_ops sn95031_vib2_dai_ops = {
 	.hw_params	= sn95031_pcm_hw_params,
 };
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 3cb3271c5fe2..620411c384e5 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -498,7 +498,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops ssm2602_dai_ops = {
+static const struct snd_soc_dai_ops ssm2602_dai_ops = {
 	.startup	= ssm2602_startup,
 	.hw_params	= ssm2602_hw_params,
 	.shutdown	= ssm2602_shutdown,
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 3b0deafd766b..e2b1cdedb982 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -783,7 +783,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static struct snd_soc_dai_ops sta32x_dai_ops = {
+static const struct snd_soc_dai_ops sta32x_dai_ops = {
 	.hw_params	= sta32x_hw_params,
 	.set_sysclk	= sta32x_set_dai_sysclk,
 	.set_fmt	= sta32x_set_dai_fmt,
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 78b2b50271e2..e4783a4f71fd 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -286,11 +286,11 @@ reset:
 	return 0;
 }
 
-static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
+static const struct snd_soc_dai_ops stac9766_dai_ops_analog = {
 	.prepare = ac97_analog_prepare,
 };
 
-static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
+static const struct snd_soc_dai_ops stac9766_dai_ops_digital = {
 	.prepare = ac97_digital_prepare,
 };
 
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 336de8f69a02..9782631df93b 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -503,7 +503,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 #define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops tlv320aic23_dai_ops = {
+static const struct snd_soc_dai_ops tlv320aic23_dai_ops = {
 	.prepare	= tlv320aic23_pcm_prepare,
 	.hw_params	= tlv320aic23_hw_params,
 	.shutdown	= tlv320aic23_shutdown,
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 7859bdcc93db..86d1fa38ed2e 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -275,7 +275,7 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 #define AIC26_FORMATS	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
 			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
 
-static struct snd_soc_dai_ops aic26_dai_ops = {
+static const struct snd_soc_dai_ops aic26_dai_ops = {
 	.hw_params	= aic26_hw_params,
 	.digital_mute	= aic26_mute,
 	.set_sysclk	= aic26_set_sysclk,
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index b21c610051c0..d2e38af46aa1 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -597,7 +597,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
 #define AIC32X4_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
 			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops aic32x4_ops = {
+static const struct snd_soc_dai_ops aic32x4_ops = {
 	.hw_params = aic32x4_hw_params,
 	.digital_mute = aic32x4_mute,
 	.set_fmt = aic32x4_set_dai_fmt,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 2e2bf18253c8..7d665ea3ac62 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1244,7 +1244,7 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed);
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops aic3x_dai_ops = {
+static const struct snd_soc_dai_ops aic3x_dai_ops = {
 	.hw_params	= aic3x_hw_params,
 	.digital_mute	= aic3x_mute,
 	.set_sysclk	= aic3x_set_dai_sysclk,
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index dc8a2b2bdc1c..abcb97e03405 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1499,7 +1499,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = {
 			 SNDRV_PCM_RATE_48000)
 #define DAC33_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops dac33_dai_ops = {
+static const struct snd_soc_dai_ops dac33_dai_ops = {
 	.startup	= dac33_startup,
 	.shutdown	= dac33_shutdown,
 	.hw_params	= dac33_hw_params,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index f798247ac1b2..2a3a52838e9c 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -2149,7 +2149,7 @@ static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate)
 #define TWL4030_RATES	 (SNDRV_PCM_RATE_8000_48000)
 #define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops twl4030_dai_hifi_ops = {
+static const struct snd_soc_dai_ops twl4030_dai_hifi_ops = {
 	.startup	= twl4030_startup,
 	.shutdown	= twl4030_shutdown,
 	.hw_params	= twl4030_hw_params,
@@ -2158,7 +2158,7 @@ static struct snd_soc_dai_ops twl4030_dai_hifi_ops = {
 	.set_tristate	= twl4030_set_tristate,
 };
 
-static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
+static const struct snd_soc_dai_ops twl4030_dai_voice_ops = {
 	.startup	= twl4030_voice_startup,
 	.shutdown	= twl4030_voice_shutdown,
 	.hw_params	= twl4030_voice_hw_params,
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 73e11f022ded..17930edd3a2c 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1397,7 +1397,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static struct snd_soc_dai_ops twl6040_dai_ops = {
+static const struct snd_soc_dai_ops twl6040_dai_ops = {
 	.startup	= twl6040_startup,
 	.hw_params	= twl6040_hw_params,
 	.prepare	= twl6040_prepare,
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index a7b8f301bad3..486aef637eed 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -452,7 +452,7 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
 SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
 };
 
-static struct snd_soc_dai_ops uda134x_dai_ops = {
+static const struct snd_soc_dai_ops uda134x_dai_ops = {
 	.startup	= uda134x_startup,
 	.shutdown	= uda134x_shutdown,
 	.hw_params	= uda134x_hw_params,
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index c5ca8cfea60f..6b933efc7ed3 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -643,21 +643,21 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 		       SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		       SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
-static struct snd_soc_dai_ops uda1380_dai_ops = {
+static const struct snd_soc_dai_ops uda1380_dai_ops = {
 	.hw_params	= uda1380_pcm_hw_params,
 	.shutdown	= uda1380_pcm_shutdown,
 	.trigger	= uda1380_trigger,
 	.set_fmt	= uda1380_set_dai_fmt_both,
 };
 
-static struct snd_soc_dai_ops uda1380_dai_ops_playback = {
+static const struct snd_soc_dai_ops uda1380_dai_ops_playback = {
 	.hw_params	= uda1380_pcm_hw_params,
 	.shutdown	= uda1380_pcm_shutdown,
 	.trigger	= uda1380_trigger,
 	.set_fmt	= uda1380_set_dai_fmt_playback,
 };
 
-static struct snd_soc_dai_ops uda1380_dai_ops_capture = {
+static const struct snd_soc_dai_ops uda1380_dai_ops_capture = {
 	.hw_params	= uda1380_pcm_hw_params,
 	.shutdown	= uda1380_pcm_shutdown,
 	.trigger	= uda1380_trigger,
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index a85498982991..9531c35dccad 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -386,7 +386,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops wl1273_dai_ops = {
+static const struct snd_soc_dai_ops wl1273_dai_ops = {
 	.startup	= wl1273_startup,
 	.hw_params	= wl1273_hw_params,
 };
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index f37d67f4058b..6c79d97ba181 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1661,7 +1661,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops wm5100_dai_ops = {
+static const struct snd_soc_dai_ops wm5100_dai_ops = {
 	.set_fmt = wm5100_set_fmt,
 	.hw_params = wm5100_hw_params,
 };
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 35f3ad83dfb6..3b846c95f07f 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1511,7 +1511,7 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect);
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8350_dai_ops = {
+static const struct snd_soc_dai_ops wm8350_dai_ops = {
 	 .hw_params	= wm8350_pcm_hw_params,
 	 .digital_mute	= wm8350_mute,
 	 .trigger	= wm8350_pcm_trigger,
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 585def1ffca6..07d84a86e14e 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1316,7 +1316,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 #define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8400_dai_ops = {
+static const struct snd_soc_dai_ops wm8400_dai_ops = {
 	.hw_params = wm8400_hw_params,
 	.digital_mute = wm8400_mute,
 	.set_fmt = wm8400_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 07c9cc759e97..26571b25e440 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -509,7 +509,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8510_dai_ops = {
+static const struct snd_soc_dai_ops wm8510_dai_ops = {
 	.hw_params	= wm8510_pcm_hw_params,
 	.digital_mute	= wm8510_mute,
 	.set_fmt	= wm8510_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index db7a6819499f..d0ae82d2b24f 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -365,7 +365,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec,
 #define WM8523_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8523_dai_ops = {
+static const struct snd_soc_dai_ops wm8523_dai_ops = {
 	.startup	= wm8523_startup,
 	.hw_params	= wm8523_hw_params,
 	.set_sysclk	= wm8523_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 8212b3c8bfdd..0aa3e4d138f4 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -776,7 +776,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8580_dai_ops_playback = {
+static const struct snd_soc_dai_ops wm8580_dai_ops_playback = {
 	.set_sysclk	= wm8580_set_sysclk,
 	.hw_params	= wm8580_paif_hw_params,
 	.set_fmt	= wm8580_set_paif_dai_fmt,
@@ -785,7 +785,7 @@ static struct snd_soc_dai_ops wm8580_dai_ops_playback = {
 	.digital_mute	= wm8580_digital_mute,
 };
 
-static struct snd_soc_dai_ops wm8580_dai_ops_capture = {
+static const struct snd_soc_dai_ops wm8580_dai_ops_capture = {
 	.set_sysclk	= wm8580_set_sysclk,
 	.hw_params	= wm8580_paif_hw_params,
 	.set_fmt	= wm8580_set_paif_dai_fmt,
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 076bdb9930a1..a6f1e391314d 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -318,7 +318,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec,
 #define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8711_ops = {
+static const struct snd_soc_dai_ops wm8711_ops = {
 	.prepare = wm8711_pcm_prepare,
 	.hw_params = wm8711_hw_params,
 	.shutdown = wm8711_shutdown,
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 04b027efd5c0..085c2f81d8c2 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -196,7 +196,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
 #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8728_dai_ops = {
+static const struct snd_soc_dai_ops wm8728_dai_ops = {
 	.hw_params	= wm8728_hw_params,
 	.digital_mute	= wm8728_mute,
 	.set_fmt	= wm8728_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index ca59622e41d2..28972d875f7c 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -465,7 +465,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8731_dai_ops = {
+static const struct snd_soc_dai_ops wm8731_dai_ops = {
 	.hw_params	= wm8731_hw_params,
 	.digital_mute	= wm8731_mute,
 	.set_sysclk	= wm8731_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index f6aef58845c2..b7d661581ebf 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -521,7 +521,7 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec,
 #define WM8737_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8737_dai_ops = {
+static const struct snd_soc_dai_ops wm8737_dai_ops = {
 	.hw_params	= wm8737_hw_params,
 	.set_sysclk	= wm8737_set_dai_sysclk,
 	.set_fmt	= wm8737_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 57ad22aacc51..e51f4f0a93f4 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -382,7 +382,7 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai,
 #define WM8741_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8741_dai_ops = {
+static const struct snd_soc_dai_ops wm8741_dai_ops = {
 	.startup	= wm8741_startup,
 	.hw_params	= wm8741_hw_params,
 	.set_sysclk	= wm8741_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index ca75a8180708..dfb41ad902e1 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -643,7 +643,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8750_dai_ops = {
+static const struct snd_soc_dai_ops wm8750_dai_ops = {
 	.hw_params	= wm8750_pcm_hw_params,
 	.digital_mute	= wm8750_mute,
 	.set_fmt	= wm8750_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 13156c836c9a..fb013b152fa6 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1315,7 +1315,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
  * 3. Voice disabled - HIFI over HIFI
  * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
  */
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
+static const struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
 	.hw_params	= wm8753_i2s_hw_params,
 	.digital_mute	= wm8753_mute,
 	.set_fmt	= wm8753_hifi_set_dai_fmt,
@@ -1324,7 +1324,7 @@ static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
 	.set_sysclk	= wm8753_set_dai_sysclk,
 };
 
-static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = {
+static const struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = {
 	.hw_params	= wm8753_pcm_hw_params,
 	.digital_mute	= wm8753_mute,
 	.set_fmt	= wm8753_voice_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index aa05e6507f84..87957e862b9c 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -528,7 +528,7 @@ static int wm8770_set_bias_level(struct snd_soc_codec *codec,
 #define WM8770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8770_dai_ops = {
+static const struct snd_soc_dai_ops wm8770_dai_ops = {
 	.digital_mute = wm8770_mute,
 	.hw_params = wm8770_hw_params,
 	.set_fmt = wm8770_set_fmt,
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index f967c59dbbef..223fc5a5c1b0 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -327,14 +327,14 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
 #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8776_dac_ops = {
+static const struct snd_soc_dai_ops wm8776_dac_ops = {
 	.digital_mute	= wm8776_mute,
 	.hw_params      = wm8776_hw_params,
 	.set_fmt        = wm8776_set_fmt,
 	.set_sysclk     = wm8776_set_sysclk,
 };
 
-static struct snd_soc_dai_ops wm8776_adc_ops = {
+static const struct snd_soc_dai_ops wm8776_adc_ops = {
 	.hw_params      = wm8776_hw_params,
 	.set_fmt        = wm8776_set_fmt,
 	.set_sysclk     = wm8776_set_sysclk,
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 9ee072b85975..d99c6a0a0a2d 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -670,7 +670,7 @@ err_reg_get:
 	return ret;
 }
 
-static struct snd_soc_dai_ops wm8804_dai_ops = {
+static const struct snd_soc_dai_ops wm8804_dai_ops = {
 	.hw_params = wm8804_hw_params,
 	.set_fmt = wm8804_set_fmt,
 	.set_sysclk = wm8804_set_sysclk,
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 17a12c2df8da..a430930cc09f 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -987,7 +987,7 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
-static struct snd_soc_dai_ops wm8900_dai_ops = {
+static const struct snd_soc_dai_ops wm8900_dai_ops = {
 	.hw_params	= wm8900_hw_params,
 	.set_clkdiv	= wm8900_set_dai_clkdiv,
 	.set_pll	= wm8900_set_dai_pll,
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 4ad8ebd290e3..812dce95f131 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1732,7 +1732,7 @@ static irqreturn_t wm8903_irq(int irq, void *data)
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8903_dai_ops = {
+static const struct snd_soc_dai_ops wm8903_dai_ops = {
 	.hw_params	= wm8903_hw_params,
 	.digital_mute	= wm8903_digital_mute,
 	.set_fmt	= wm8903_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index bb070f835257..f0b0c7a487b3 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2205,7 +2205,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 #define WM8904_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8904_dai_ops = {
+static const struct snd_soc_dai_ops wm8904_dai_ops = {
 	.set_sysclk = wm8904_set_sysclk,
 	.set_fmt = wm8904_set_fmt,
 	.set_tdm_slot = wm8904_set_tdm_slot,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 1b5856b4ea7c..0dd1e0c0fc1b 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -644,7 +644,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 			SNDRV_PCM_FMTBIT_S24_LE |			\
 			SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8940_dai_ops = {
+static const struct snd_soc_dai_ops wm8940_dai_ops = {
 	.hw_params = wm8940_i2s_hw_params,
 	.set_sysclk = wm8940_set_dai_sysclk,
 	.digital_mute = wm8940_mute,
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 3c7198779c31..dbf2a8328a8e 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -859,7 +859,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec,
 #define WM8955_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8955_dai_ops = {
+static const struct snd_soc_dai_ops wm8955_dai_ops = {
 	.set_sysclk = wm8955_set_sysclk,
 	.set_fmt = wm8955_set_fmt,
 	.hw_params = wm8955_hw_params,
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 6e22f9b3d967..06dca88a7332 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -869,7 +869,7 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec,
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8960_dai_ops = {
+static const struct snd_soc_dai_ops wm8960_dai_ops = {
 	.hw_params = wm8960_hw_params,
 	.digital_mute = wm8960_mute,
 	.set_fmt = wm8960_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 7f2df7ba27f6..783a3d1daf51 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -929,7 +929,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8961_dai_ops = {
+static const struct snd_soc_dai_ops wm8961_dai_ops = {
 	.hw_params = wm8961_hw_params,
 	.set_sysclk = wm8961_set_sysclk,
 	.set_fmt = wm8961_set_fmt,
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 48b5c95a0648..555311d1ce37 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3503,7 +3503,7 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute)
 #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8962_dai_ops = {
+static const struct snd_soc_dai_ops wm8962_dai_ops = {
 	.hw_params = wm8962_hw_params,
 	.set_sysclk = wm8962_set_dai_sysclk,
 	.set_fmt = wm8962_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 3a06a95dd96f..98bfbdd62c60 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -567,7 +567,7 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
 #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8971_dai_ops = {
+static const struct snd_soc_dai_ops wm8971_dai_ops = {
 	.hw_params	= wm8971_pcm_hw_params,
 	.digital_mute	= wm8971_mute,
 	.set_fmt	= wm8971_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 7bd35b8fdcd2..16569c7a03c1 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -557,7 +557,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec,
 #define WM8974_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8974_ops = {
+static const struct snd_soc_dai_ops wm8974_ops = {
 	.hw_params = wm8974_pcm_hw_params,
 	.digital_mute = wm8974_mute,
 	.set_fmt = wm8974_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 41ca4d9ac20c..517bb2238d46 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -865,7 +865,7 @@ static int wm8978_set_bias_level(struct snd_soc_codec *codec,
 #define WM8978_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8978_dai_ops = {
+static const struct snd_soc_dai_ops wm8978_dai_ops = {
 	.hw_params	= wm8978_hw_params,
 	.digital_mute	= wm8978_mute,
 	.set_fmt	= wm8978_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index 58e067b5a6a3..362298cce92c 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -1035,7 +1035,7 @@ static int wm8983_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_dai_ops wm8983_dai_ops = {
+static const struct snd_soc_dai_ops wm8983_dai_ops = {
 	.digital_mute = wm8983_dac_mute,
 	.hw_params = wm8983_hw_params,
 	.set_fmt = wm8983_set_fmt,
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 36c4ee08e159..9e4481bb1223 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -1031,7 +1031,7 @@ err_reg_get:
 	return ret;
 }
 
-static struct snd_soc_dai_ops wm8985_dai_ops = {
+static const struct snd_soc_dai_ops wm8985_dai_ops = {
 	.digital_mute = wm8985_dac_mute,
 	.hw_params = wm8985_hw_params,
 	.set_fmt = wm8985_set_fmt,
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 514189d1923e..9d83bed5c210 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -701,7 +701,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec,
 #define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8988_ops = {
+static const struct snd_soc_dai_ops wm8988_ops = {
 	.startup = wm8988_pcm_startup,
 	.hw_params = wm8988_pcm_hw_params,
 	.set_fmt = wm8988_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index d4cbec6372db..61c620e5fe4f 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1287,7 +1287,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
  * 1. ADC/DAC on Primary Interface
  * 2. ADC on Primary Interface/DAC on secondary
  */
-static struct snd_soc_dai_ops wm8990_dai_ops = {
+static const struct snd_soc_dai_ops wm8990_dai_ops = {
 	.hw_params	= wm8990_hw_params,
 	.digital_mute	= wm8990_mute,
 	.set_fmt	= wm8990_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 1d46d59c82a3..ac957ece6785 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1311,7 +1311,7 @@ static int wm8991_probe(struct snd_soc_codec *codec)
 #define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops wm8991_ops = {
+static const struct snd_soc_dai_ops wm8991_ops = {
 	.hw_params = wm8991_hw_params,
 	.digital_mute = wm8991_mute,
 	.set_fmt = wm8991_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index d1a142f48b09..780c24cdab6d 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1394,7 +1394,7 @@ out:
 	return 0;
 }
 
-static struct snd_soc_dai_ops wm8993_ops = {
+static const struct snd_soc_dai_ops wm8993_ops = {
 	.set_sysclk = wm8993_set_sysclk,
 	.set_fmt = wm8993_set_dai_fmt,
 	.hw_params = wm8993_hw_params,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 9c982e47eb99..73db9806c475 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2531,7 +2531,7 @@ static int wm8994_aif2_probe(struct snd_soc_dai *dai)
 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
+static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
 	.set_sysclk	= wm8994_set_dai_sysclk,
 	.set_fmt	= wm8994_set_dai_fmt,
 	.hw_params	= wm8994_hw_params,
@@ -2541,7 +2541,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
 	.set_tristate	= wm8994_set_tristate,
 };
 
-static struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
+static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
 	.set_sysclk	= wm8994_set_dai_sysclk,
 	.set_fmt	= wm8994_set_dai_fmt,
 	.hw_params	= wm8994_hw_params,
@@ -2551,7 +2551,7 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
 	.set_tristate	= wm8994_set_tristate,
 };
 
-static struct snd_soc_dai_ops wm8994_aif3_dai_ops = {
+static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = {
 	.hw_params	= wm8994_aif3_hw_params,
 	.set_tristate	= wm8994_set_tristate,
 };
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 3774acb69ddd..8f6a36d7c75b 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -2155,7 +2155,7 @@ err_reg_get:
 #define WM8995_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8995_aif1_dai_ops = {
+static const struct snd_soc_dai_ops wm8995_aif1_dai_ops = {
 	.set_sysclk = wm8995_set_dai_sysclk,
 	.set_fmt = wm8995_set_dai_fmt,
 	.hw_params = wm8995_hw_params,
@@ -2164,7 +2164,7 @@ static struct snd_soc_dai_ops wm8995_aif1_dai_ops = {
 	.set_tristate = wm8995_set_tristate,
 };
 
-static struct snd_soc_dai_ops wm8995_aif2_dai_ops = {
+static const struct snd_soc_dai_ops wm8995_aif2_dai_ops = {
 	.set_sysclk = wm8995_set_dai_sysclk,
 	.set_fmt = wm8995_set_dai_fmt,
 	.hw_params = wm8995_hw_params,
@@ -2173,7 +2173,7 @@ static struct snd_soc_dai_ops wm8995_aif2_dai_ops = {
 	.set_tristate = wm8995_set_tristate,
 };
 
-static struct snd_soc_dai_ops wm8995_aif3_dai_ops = {
+static const struct snd_soc_dai_ops wm8995_aif3_dai_ops = {
 	.set_tristate = wm8995_set_tristate,
 };
 
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index fd5bb1ad6912..304a0e570cb4 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -3052,7 +3052,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = {
 			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\
 			SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm8996_dai_ops = {
+static const struct snd_soc_dai_ops wm8996_dai_ops = {
 	.set_fmt = wm8996_set_fmt,
 	.hw_params = wm8996_hw_params,
 	.set_sysclk = wm8996_set_sysclk,
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index f7c0738a9da6..48bf80baf1d4 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1234,7 +1234,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 	 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops wm9081_dai_ops = {
+static const struct snd_soc_dai_ops wm9081_dai_ops = {
 	.hw_params = wm9081_hw_params,
 	.set_fmt = wm9081_set_dai_fmt,
 	.digital_mute = wm9081_digital_mute,
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 646b58dda849..edf603281ce7 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -258,7 +258,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 			SNDRV_PCM_RATE_48000)
 
-static struct snd_soc_dai_ops wm9705_dai_ops = {
+static const struct snd_soc_dai_ops wm9705_dai_ops = {
 	.prepare	= ac97_prepare,
 };
 
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 90117f8156e8..fd1812704af8 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -505,11 +505,11 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
-static struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
+static const struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
 	.prepare	= ac97_prepare,
 };
 
-static struct snd_soc_dai_ops wm9712_dai_ops_aux = {
+static const struct snd_soc_dai_ops wm9712_dai_ops_aux = {
 	.prepare	= ac97_aux_prepare,
 };
 
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 7167cb6787db..09360b60037c 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1026,19 +1026,19 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
-static struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
+static const struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
 	.prepare	= ac97_hifi_prepare,
 	.set_clkdiv	= wm9713_set_dai_clkdiv,
 	.set_pll	= wm9713_set_dai_pll,
 };
 
-static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
+static const struct snd_soc_dai_ops wm9713_dai_ops_aux = {
 	.prepare	= ac97_aux_prepare,
 	.set_clkdiv	= wm9713_set_dai_clkdiv,
 	.set_pll	= wm9713_set_dai_pll,
 };
 
-static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
+static const struct snd_soc_dai_ops wm9713_dai_ops_voice = {
 	.hw_params	= wm9713_pcm_hw_params,
 	.set_clkdiv	= wm9713_set_dai_clkdiv,
 	.set_pll	= wm9713_set_dai_pll,
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 300e12118c00..f3d5ae1078be 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -620,7 +620,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
-static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+static const struct snd_soc_dai_ops davinci_i2s_dai_ops = {
 	.startup	= davinci_i2s_startup,
 	.shutdown	= davinci_i2s_shutdown,
 	.prepare	= davinci_i2s_prepare,
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7173df254a91..03cea9d39c4b 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -813,7 +813,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
 	.startup	= davinci_mcasp_startup,
 	.trigger	= davinci_mcasp_trigger,
 	.hw_params	= davinci_mcasp_hw_params,
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 1f11525d97e8..dae96b85fd6d 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -183,7 +183,7 @@ static int davinci_vcif_startup(struct snd_pcm_substream *substream,
 
 #define DAVINCI_VCIF_RATES	SNDRV_PCM_RATE_8000_48000
 
-static struct snd_soc_dai_ops davinci_vcif_dai_ops = {
+static const struct snd_soc_dai_ops davinci_vcif_dai_ops = {
 	.startup	= davinci_vcif_startup,
 	.trigger	= davinci_vcif_trigger,
 	.hw_params	= davinci_vcif_hw_params,
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 3cd6158d83e1..c423d12a26cf 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -330,7 +330,7 @@ static int ep93xx_ac97_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops ep93xx_ac97_dai_ops = {
+static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = {
 	.startup	= ep93xx_ac97_startup,
 	.trigger	= ep93xx_ac97_trigger,
 };
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 099614e16651..3dba128cc6f1 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -338,7 +338,7 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
 #define ep93xx_i2s_resume	NULL
 #endif
 
-static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = {
+static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = {
 	.startup	= ep93xx_i2s_startup,
 	.shutdown	= ep93xx_i2s_shutdown,
 	.hw_params	= ep93xx_i2s_hw_params,
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 83c4bd5b2dd7..17d857e55efe 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -514,7 +514,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
 	}
 }
 
-static struct snd_soc_dai_ops fsl_ssi_dai_ops = {
+static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
 	.startup	= fsl_ssi_startup,
 	.hw_params	= fsl_ssi_hw_params,
 	.shutdown	= fsl_ssi_shutdown,
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index ad36b095bb79..2fb388f0150b 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -226,12 +226,12 @@ static int psc_ac97_probe(struct snd_soc_dai *cpu_dai)
 /**
  * psc_ac97_dai_template: template CPU Digital Audio Interface
  */
-static struct snd_soc_dai_ops psc_ac97_analog_ops = {
+static const struct snd_soc_dai_ops psc_ac97_analog_ops = {
 	.hw_params	= psc_ac97_hw_analog_params,
 	.trigger	= psc_ac97_trigger,
 };
 
-static struct snd_soc_dai_ops psc_ac97_digital_ops = {
+static const struct snd_soc_dai_ops psc_ac97_digital_ops = {
 	.hw_params	= psc_ac97_hw_digital_params,
 };
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 87cf2a5c2b2c..e77a1f20d4d2 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -123,7 +123,7 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
-static struct snd_soc_dai_ops psc_i2s_dai_ops = {
+static const struct snd_soc_dai_ops psc_i2s_dai_ops = {
 	.hw_params	= psc_i2s_hw_params,
 	.set_sysclk	= psc_i2s_set_sysclk,
 	.set_fmt	= psc_i2s_set_fmt,
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 4c05e2b8f4d2..eed7041364e6 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -342,7 +342,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
-static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = {
+static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = {
 	.hw_params	= imx_ssi_hw_params,
 	.set_fmt	= imx_ssi_set_dai_fmt,
 	.set_clkdiv	= imx_ssi_set_dai_clkdiv,
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index cd22a54b2f14..91255c6e1ee7 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -392,7 +392,7 @@ static int jz4740_i2s_dai_remove(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops jz4740_i2s_dai_ops = {
+static const struct snd_soc_dai_ops jz4740_i2s_dai_ops = {
 	.startup = jz4740_i2s_startup,
 	.shutdown = jz4740_i2s_shutdown,
 	.trigger = jz4740_i2s_trigger,
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 715e841c0507..2b212dcb9ac7 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -373,7 +373,7 @@ static int kirkwood_i2s_remove(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops kirkwood_i2s_dai_ops = {
+static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = {
 	.startup	= kirkwood_i2s_startup,
 	.trigger	= kirkwood_i2s_trigger,
 	.hw_params      = kirkwood_i2s_hw_params,
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 76dc74d24fc2..46d76b52529b 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -550,7 +550,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops mxs_saif_dai_ops = {
+static const struct snd_soc_dai_ops mxs_saif_dai_ops = {
 	.startup = mxs_saif_startup,
 	.trigger = mxs_saif_trigger,
 	.prepare = mxs_saif_prepare,
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 9c0edad90d8b..7544d249807e 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -291,7 +291,7 @@ static int nuc900_ac97_remove(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
+static const struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
 	.trigger	= nuc900_ac97_trigger,
 };
 
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index ccb8a6aa1817..a04a4338fdac 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -474,7 +474,7 @@ static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
 }
 
 /* Our codec DAI probably doesn't have its own .ops structure */
-static struct snd_soc_dai_ops ams_delta_dai_ops = {
+static const struct snd_soc_dai_ops ams_delta_dai_ops = {
 	.digital_mute = ams_delta_digital_mute,
 };
 
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
index 36c6eaeffb02..9bb1cf89b4a4 100644
--- a/sound/soc/omap/omap-hdmi.c
+++ b/sound/soc/omap/omap-hdmi.c
@@ -83,7 +83,7 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
 	return err;
 }
 
-static struct snd_soc_dai_ops omap_hdmi_dai_ops = {
+static const struct snd_soc_dai_ops omap_hdmi_dai_ops = {
 	.startup	= omap_hdmi_dai_startup,
 	.hw_params	= omap_hdmi_dai_hw_params,
 };
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 4314647e735e..d91e6efd2600 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -599,7 +599,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
-static struct snd_soc_dai_ops mcbsp_dai_ops = {
+static const struct snd_soc_dai_ops mcbsp_dai_ops = {
 	.startup	= omap_mcbsp_dai_startup,
 	.shutdown	= omap_mcbsp_dai_shutdown,
 	.trigger	= omap_mcbsp_dai_trigger,
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 41d17067cc73..cc8ceff25dbd 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -367,7 +367,7 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops omap_mcpdm_dai_ops = {
+static const struct snd_soc_dai_ops omap_mcpdm_dai_ops = {
 	.startup	= omap_mcpdm_dai_startup,
 	.shutdown	= omap_mcpdm_dai_shutdown,
 	.hw_params	= omap_mcpdm_dai_hw_params,
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 8ad93ee2e92b..9c9a51ef67c3 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -771,7 +771,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai)
 			    SNDRV_PCM_FMTBIT_S24_LE |	\
 			    SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
 	.startup	= pxa_ssp_startup,
 	.shutdown	= pxa_ssp_shutdown,
 	.trigger	= pxa_ssp_trigger,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index ac51c6d25c42..3fec2f35b8f8 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -163,15 +163,15 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000)
 
-static struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
+static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
 	.hw_params	= pxa2xx_ac97_hw_params,
 };
 
-static struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
+static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
 	.hw_params	= pxa2xx_ac97_hw_aux_params,
 };
 
-static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
+static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
 	.hw_params	= pxa2xx_ac97_hw_mic_params,
 };
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 11be5952a506..609abd51e55f 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -331,7 +331,7 @@ static int  pxa2xx_i2s_remove(struct snd_soc_dai *dai)
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
 
-static struct snd_soc_dai_ops pxa_i2s_dai_ops = {
+static const struct snd_soc_dai_ops pxa_i2s_dai_ops = {
 	.startup	= pxa2xx_i2s_startup,
 	.shutdown	= pxa2xx_i2s_shutdown,
 	.trigger	= pxa2xx_i2s_trigger,
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 3052f64b2403..13716a9317fb 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -409,7 +409,7 @@ static int s6000_i2s_dai_probe(struct snd_soc_dai *dai)
 			 SNDRV_PCM_RATE_8000_192000)
 #define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops s6000_i2s_dai_ops = {
+static const struct snd_soc_dai_ops s6000_i2s_dai_ops = {
 	.set_fmt = s6000_i2s_set_dai_fmt,
 	.set_clkdiv = s6000_i2s_set_clkdiv,
 	.hw_params = s6000_i2s_hw_params,
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 16521e3ffc0c..09035afdeb74 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -329,12 +329,12 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops s3c_ac97_dai_ops = {
+static const struct snd_soc_dai_ops s3c_ac97_dai_ops = {
 	.hw_params	= s3c_ac97_hw_params,
 	.trigger	= s3c_ac97_trigger,
 };
 
-static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = {
+static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = {
 	.hw_params	= s3c_ac97_hw_mic_params,
 	.trigger	= s3c_ac97_mic_trigger,
 };
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index bff42bf370b9..03ee8ce46a29 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -923,7 +923,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops samsung_i2s_dai_ops = {
+static const struct snd_soc_dai_ops samsung_i2s_dai_ops = {
 	.trigger = i2s_trigger,
 	.hw_params = i2s_hw_params,
 	.set_fmt = i2s_set_fmt,
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 05a47cf7f06e..2df2762f3000 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -452,7 +452,7 @@ static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
-static struct snd_soc_dai_ops s3c_pcm_dai_ops = {
+static const struct snd_soc_dai_ops s3c_pcm_dai_ops = {
 	.set_sysclk	= s3c_pcm_set_sysclk,
 	.set_clkdiv	= s3c_pcm_set_clkdiv,
 	.trigger	= s3c_pcm_trigger,
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 7bbec25e6e15..545773d0641c 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -142,7 +142,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
-static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
+static const struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
 	.hw_params	= s3c2412_i2s_hw_params,
 };
 
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 558c64bbed2e..2a98bed2db02 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -444,7 +444,7 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
-static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = {
+static const struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = {
 	.trigger	= s3c24xx_i2s_trigger,
 	.hw_params	= s3c24xx_i2s_hw_params,
 	.set_fmt	= s3c24xx_i2s_set_fmt,
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index 468cff1bb1af..a1fee1a414c9 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -334,7 +334,7 @@ static int spdif_resume(struct snd_soc_dai *cpu_dai)
 #define spdif_resume NULL
 #endif
 
-static struct snd_soc_dai_ops spdif_dai_ops = {
+static const struct snd_soc_dai_ops spdif_dai_ops = {
 	.set_sysclk	= spdif_set_sysclk,
 	.trigger	= spdif_trigger,
 	.hw_params	= spdif_hw_params,
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 99ed61024166..aa3033075a0d 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1096,7 +1096,7 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static struct snd_soc_dai_ops fsi_dai_ops = {
+static const struct snd_soc_dai_ops fsi_dai_ops = {
 	.startup	= fsi_dai_startup,
 	.shutdown	= fsi_dai_shutdown,
 	.trigger	= fsi_dai_trigger,
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index c87e3ff28a0a..a1f307b9a82d 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream,
 #define AC97_FMTS	\
 	SNDRV_PCM_FMTBIT_S16_LE
 
-static struct snd_soc_dai_ops hac_dai_ops = {
+static const struct snd_soc_dai_ops hac_dai_ops = {
 	.hw_params	= hac_hw_params,
 };
 
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index edacfeb13b94..93dea49ff1a7 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -707,7 +707,7 @@ epclkget:
 	return ret;
 }
 
-static struct snd_soc_dai_ops siu_dai_ops = {
+static const struct snd_soc_dai_ops siu_dai_ops = {
 	.startup	= siu_dai_startup,
 	.shutdown	= siu_dai_shutdown,
 	.prepare	= siu_dai_prepare,
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index e0c621c0553b..1fda16a00e6a 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
 	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
 
-static struct snd_soc_dai_ops ssi_dai_ops = {
+static const struct snd_soc_dai_ops ssi_dai_ops = {
 	.startup	= ssi_startup,
 	.shutdown	= ssi_shutdown,
 	.trigger	= ssi_trigger,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a5d3685a5d38..bf41d9071f1e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -735,7 +735,7 @@ EXPORT_SYMBOL_GPL(snd_soc_resume);
 #define snd_soc_resume NULL
 #endif
 
-static struct snd_soc_dai_ops null_dai_ops = {
+static const struct snd_soc_dai_ops null_dai_ops = {
 };
 
 static int soc_bind_dai_link(struct snd_soc_card *card, int num)
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 76014f0d8a29..1acbb5541772 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -305,7 +305,7 @@ static int tegra_i2s_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops tegra_i2s_dai_ops = {
+static const struct snd_soc_dai_ops tegra_i2s_dai_ops = {
 	.set_fmt	= tegra_i2s_set_fmt,
 	.hw_params	= tegra_i2s_hw_params,
 	.trigger	= tegra_i2s_trigger,
diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
index dd11d0c63474..ea9c92036aa1 100644
--- a/sound/soc/tegra/tegra_spdif.c
+++ b/sound/soc/tegra/tegra_spdif.c
@@ -226,7 +226,7 @@ static int tegra_spdif_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops tegra_spdif_dai_ops = {
+static const struct snd_soc_dai_ops tegra_spdif_dai_ops = {
 	.hw_params	= tegra_spdif_hw_params,
 	.trigger	= tegra_spdif_trigger,
 };
-- 
cgit v1.2.3


From 186bcda6f6217dc4b5353c3474121bc1194847f6 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 Nov 2011 18:21:18 -0700
Subject: ASoC: Tegra DAS: Add device tree binding

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 Documentation/devicetree/bindings/sound/tegra20-das.txt | 12 ++++++++++++
 sound/soc/tegra/tegra_das.c                             |  8 ++++++++
 2 files changed, 20 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/tegra20-das.txt

(limited to 'sound/soc/tegra')

diff --git a/Documentation/devicetree/bindings/sound/tegra20-das.txt b/Documentation/devicetree/bindings/sound/tegra20-das.txt
new file mode 100644
index 000000000000..6de3a7ee4efb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra20-das.txt
@@ -0,0 +1,12 @@
+NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
+
+Required properties:
+- compatible : "nvidia,tegra20-das"
+- reg : Should contain DAS registers location and length
+
+Example:
+
+das@70000c00 {
+	compatible = "nvidia,tegra20-das";
+	reg = <0x70000c00 0x80>;
+};
diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c
index fa3a4426cbdd..5b82b4e79231 100644
--- a/sound/soc/tegra/tegra_das.c
+++ b/sound/soc/tegra/tegra_das.c
@@ -225,11 +225,18 @@ static int __devexit tegra_das_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct of_device_id tegra_das_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra20-das", },
+	{},
+};
+
 static struct platform_driver tegra_das_driver = {
 	.probe = tegra_das_probe,
 	.remove = __devexit_p(tegra_das_remove),
 	.driver = {
 		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.of_match_table = tegra_das_of_match,
 	},
 };
 module_platform_driver(tegra_das_driver);
@@ -238,3 +245,4 @@ MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra DAS driver");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra_das_of_match);
-- 
cgit v1.2.3


From e4e4c18a930ff11940ba2c525676566bd631706f Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 Nov 2011 18:21:20 -0700
Subject: ASoC: Tegra+WM8903 machine: Use devm_ APIs and module_platform_driver

module_platform_driver saves some boiler-plate code.

The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 26 +++++++-------------------
 1 file changed, 7 insertions(+), 19 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 9b0ee1510935..33feee81668c 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -390,17 +390,19 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	machine = kzalloc(sizeof(struct tegra_wm8903), GFP_KERNEL);
+	machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8903),
+			       GFP_KERNEL);
 	if (!machine) {
 		dev_err(&pdev->dev, "Can't allocate tegra_wm8903 struct\n");
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto err;
 	}
 
 	machine->pdata = pdata;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
-		goto err_free_machine;
+		goto err;
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
@@ -431,8 +433,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
-err_free_machine:
-	kfree(machine);
+err:
 	return ret;
 }
 
@@ -460,8 +461,6 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
-	kfree(machine);
-
 	return 0;
 }
 
@@ -474,18 +473,7 @@ static struct platform_driver tegra_wm8903_driver = {
 	.probe = tegra_wm8903_driver_probe,
 	.remove = __devexit_p(tegra_wm8903_driver_remove),
 };
-
-static int __init tegra_wm8903_modinit(void)
-{
-	return platform_driver_register(&tegra_wm8903_driver);
-}
-module_init(tegra_wm8903_modinit);
-
-static void __exit tegra_wm8903_modexit(void)
-{
-	platform_driver_unregister(&tegra_wm8903_driver);
-}
-module_exit(tegra_wm8903_modexit);
+module_platform_driver(tegra_wm8903_driver);
 
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra+WM8903 machine ASoC driver");
-- 
cgit v1.2.3


From 45c26091205eb6ad737329c5973f46fd7c122595 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 Nov 2011 18:21:21 -0700
Subject: ASoC: Tegra TrimSlice machine: Use devm_ APIs and
 module_platform_driver

module_platform_driver saves some boiler-plate code.

The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/trimslice.c | 26 +++++++-------------------
 1 file changed, 7 insertions(+), 19 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 2699a6fa45f9..d564b40756a9 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -170,15 +170,17 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
 	struct tegra_trimslice *trimslice;
 	int ret;
 
-	trimslice = kzalloc(sizeof(struct tegra_trimslice), GFP_KERNEL);
+	trimslice = devm_kzalloc(&pdev->dev, sizeof(struct tegra_trimslice),
+				 GFP_KERNEL);
 	if (!trimslice) {
 		dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n");
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto err;
 	}
 
 	ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev);
 	if (ret)
-		goto err_free_trimslice;
+		goto err;
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
@@ -195,8 +197,7 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&trimslice->util_data);
-err_free_trimslice:
-	kfree(trimslice);
+err:
 	return ret;
 }
 
@@ -209,8 +210,6 @@ static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&trimslice->util_data);
 
-	kfree(trimslice);
-
 	return 0;
 }
 
@@ -222,18 +221,7 @@ static struct platform_driver tegra_snd_trimslice_driver = {
 	.probe = tegra_snd_trimslice_probe,
 	.remove = __devexit_p(tegra_snd_trimslice_remove),
 };
-
-static int __init snd_tegra_trimslice_init(void)
-{
-	return platform_driver_register(&tegra_snd_trimslice_driver);
-}
-module_init(snd_tegra_trimslice_init);
-
-static void __exit snd_tegra_trimslice_exit(void)
-{
-	platform_driver_unregister(&tegra_snd_trimslice_driver);
-}
-module_exit(snd_tegra_trimslice_exit);
+module_platform_driver(tegra_snd_trimslice_driver);
 
 MODULE_AUTHOR("Mike Rapoport <mike@compulab.co.il>");
 MODULE_DESCRIPTION("Trimslice machine ASoC driver");
-- 
cgit v1.2.3


From d4a2eca781bfd7323bfd98dbc7fd63c7d613fef2 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 23 Nov 2011 13:33:25 -0700
Subject: ASoC: Tegra I2S: Remove dependency on pdev->id

When devices are instantiated from device-tree, pdev->id is set to -1.
Rework the driver so it doesn't depend on the ID.

Tegra I2S instantiated from board files are configured with pdev
name "tegra-i2s" and ID 0 or 1. The driver core then names the
device "tegra-i2s.0" or "tegra-i2s.1". This is not changing.

When a device is instantiated from device-tree, it will have
pdev->name="" and pdev->id=-1. For this reason, the pdev->id value is
not something we can rely on.

This patch doesn't actually change any names though:

When a device is instantiated from device-tree, the overall device name
will be "${unit_address}.${node_name}". This causes issues such as
clk_get() failures due to lack of a device-name match. To solve that,
AUXDATA was invented, to force a specific device name, thus allowing
dev_name() to return the same as the non-device-tree case. Tegra
currently uses AUXDATA for the I2S controllers. Eventually, AUXDATA will
go away, most likely replaced by phandle-based references within the
device tree.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_i2s.c | 72 +++++++++++++++------------------------------
 sound/soc/tegra/tegra_i2s.h |  1 +
 2 files changed, 24 insertions(+), 49 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 1acbb5541772..ca4d0c0a913e 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -98,13 +98,11 @@ static const struct file_operations tegra_i2s_debug_fops = {
 	.release = single_release,
 };
 
-static void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id)
+static void tegra_i2s_debug_add(struct tegra_i2s *i2s)
 {
-	char name[] = DRV_NAME ".0";
-
-	snprintf(name, sizeof(name), DRV_NAME".%1d", id);
-	i2s->debug = debugfs_create_file(name, S_IRUGO, snd_soc_debugfs_root,
-						i2s, &tegra_i2s_debug_fops);
+	i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO,
+					 snd_soc_debugfs_root, i2s,
+					 &tegra_i2s_debug_fops);
 }
 
 static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
@@ -311,43 +309,22 @@ static const struct snd_soc_dai_ops tegra_i2s_dai_ops = {
 	.trigger	= tegra_i2s_trigger,
 };
 
-static struct snd_soc_dai_driver tegra_i2s_dai[] = {
-	{
-		.name = DRV_NAME ".0",
-		.probe = tegra_i2s_probe,
-		.playback = {
-			.channels_min = 2,
-			.channels_max = 2,
-			.rates = SNDRV_PCM_RATE_8000_96000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
-		},
-		.capture = {
-			.channels_min = 2,
-			.channels_max = 2,
-			.rates = SNDRV_PCM_RATE_8000_96000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
-		},
-		.ops = &tegra_i2s_dai_ops,
-		.symmetric_rates = 1,
+static const struct snd_soc_dai_driver tegra_i2s_dai_template = {
+	.probe = tegra_i2s_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	{
-		.name = DRV_NAME ".1",
-		.probe = tegra_i2s_probe,
-		.playback = {
-			.channels_min = 2,
-			.channels_max = 2,
-			.rates = SNDRV_PCM_RATE_8000_96000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
-		},
-		.capture = {
-			.channels_min = 2,
-			.channels_max = 2,
-			.rates = SNDRV_PCM_RATE_8000_96000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
-		},
-		.ops = &tegra_i2s_dai_ops,
-		.symmetric_rates = 1,
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
+	.ops = &tegra_i2s_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
@@ -356,12 +333,6 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 	struct resource *mem, *memregion, *dmareq;
 	int ret;
 
-	if ((pdev->id < 0) ||
-		(pdev->id >= ARRAY_SIZE(tegra_i2s_dai))) {
-		dev_err(&pdev->dev, "ID %d out of range\n", pdev->id);
-		return -EINVAL;
-	}
-
 	i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL);
 	if (!i2s) {
 		dev_err(&pdev->dev, "Can't allocate tegra_i2s\n");
@@ -370,6 +341,9 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 	}
 	dev_set_drvdata(&pdev->dev, i2s);
 
+	i2s->dai = tegra_i2s_dai_template;
+	i2s->dai.name = dev_name(&pdev->dev);
+
 	i2s->clk_i2s = clk_get(&pdev->dev, NULL);
 	if (IS_ERR(i2s->clk_i2s)) {
 		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
@@ -418,14 +392,14 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 
 	i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED;
 
-	ret = snd_soc_register_dai(&pdev->dev, &tegra_i2s_dai[pdev->id]);
+	ret = snd_soc_register_dai(&pdev->dev, &i2s->dai);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
 		ret = -ENOMEM;
 		goto err_clk_put;
 	}
 
-	tegra_i2s_debug_add(i2s, pdev->id);
+	tegra_i2s_debug_add(i2s);
 
 	return 0;
 
diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h
index 2b38a096f46c..15ce1e2e8bde 100644
--- a/sound/soc/tegra/tegra_i2s.h
+++ b/sound/soc/tegra/tegra_i2s.h
@@ -153,6 +153,7 @@
 #define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
 
 struct tegra_i2s {
+	struct snd_soc_dai_driver dai;
 	struct clk *clk_i2s;
 	int clk_refs;
 	struct tegra_pcm_dma_params capture_dma_data;
-- 
cgit v1.2.3


From 6e5fdba9c9d4e2fdb19bf19633cb7b9bb72dccb1 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 23 Nov 2011 12:42:05 -0700
Subject: ASoC: Tegra+WM903 machine: Set the new fully_routed flag

Set card.fully_routed to request the ASoC core calculated unused codec
pins, and call snd_soc_dapm_nc_pin() for them. Remove the open-coded
calls.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 22 +---------------------
 1 file changed, 1 insertion(+), 21 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 33feee81668c..b260f54a4462 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -331,27 +331,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 
 	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
 
-	/* FIXME: Calculate automatically based on DAPM routes? */
-	if (!machine_is_harmony())
-		snd_soc_dapm_nc_pin(dapm, "IN1L");
-	if (!machine_is_seaboard() && !machine_is_aebl())
-		snd_soc_dapm_nc_pin(dapm, "IN1R");
-	snd_soc_dapm_nc_pin(dapm, "IN2L");
-	if (!machine_is_kaen())
-		snd_soc_dapm_nc_pin(dapm, "IN2R");
-	snd_soc_dapm_nc_pin(dapm, "IN3L");
-	snd_soc_dapm_nc_pin(dapm, "IN3R");
-
-	if (machine_is_aebl()) {
-		snd_soc_dapm_nc_pin(dapm, "LON");
-		snd_soc_dapm_nc_pin(dapm, "RON");
-		snd_soc_dapm_nc_pin(dapm, "ROP");
-		snd_soc_dapm_nc_pin(dapm, "LOP");
-	} else {
-		snd_soc_dapm_nc_pin(dapm, "LINEOUTR");
-		snd_soc_dapm_nc_pin(dapm, "LINEOUTL");
-	}
-
 	return 0;
 }
 
@@ -375,6 +354,7 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = {
 	.num_controls = ARRAY_SIZE(tegra_wm8903_controls),
 	.dapm_widgets = tegra_wm8903_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(tegra_wm8903_dapm_widgets),
+	.fully_routed = true,
 };
 
 static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
-- 
cgit v1.2.3


From 504855d171f4183ac231a5ecdf0273ac249cda2b Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 23 Nov 2011 12:42:06 -0700
Subject: ASoC: TrimSlice machine: Set the new fully_routed flag

Set card.fully_routed to request the ASoC core calculated unused codec
pins, and call snd_soc_dapm_nc_pin() for them. Remove the open-coded
calls.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/trimslice.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index d564b40756a9..043eb7c7eb73 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -119,7 +119,6 @@ static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_card *card = codec->card;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1,
@@ -135,10 +134,6 @@ static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd)
 		return ret;
 	}
 
-	snd_soc_dapm_nc_pin(dapm, "LHPOUT");
-	snd_soc_dapm_nc_pin(dapm, "RHPOUT");
-	snd_soc_dapm_nc_pin(dapm, "MICIN");
-
 	return 0;
 }
 
@@ -162,6 +157,7 @@ static struct snd_soc_card snd_soc_trimslice = {
 	.num_dapm_widgets = ARRAY_SIZE(trimslice_dapm_widgets),
 	.dapm_routes = trimslice_audio_map,
 	.num_dapm_routes = ARRAY_SIZE(trimslice_audio_map),
+	.fully_routed = true,
 };
 
 static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
-- 
cgit v1.2.3


From 5032dc34294d1084b7367877dadb6edb2d45ad7c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 27 Nov 2011 12:20:08 +0000
Subject: ASoC: Convert WM8903 MICBIAS to a supply widget

Also rename it to MICBIAS to reflect the pin name and help any out of tree
users notice the change.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
---
 sound/soc/codecs/wm8903.c      |  4 ++--
 sound/soc/tegra/tegra_wm8903.c | 18 +++++++++---------
 2 files changed, 11 insertions(+), 11 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 5957a8b52eda..70a2268c5498 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -838,7 +838,7 @@ SND_SOC_DAPM_OUTPUT("LON"),
 SND_SOC_DAPM_OUTPUT("ROP"),
 SND_SOC_DAPM_OUTPUT("RON"),
 
-SND_SOC_DAPM_MICBIAS("Mic Bias", WM8903_MIC_BIAS_CONTROL_0, 0, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS", WM8903_MIC_BIAS_CONTROL_0, 0, 0, NULL, 0),
 
 SND_SOC_DAPM_MUX("Left Input Mux", SND_SOC_NOPM, 0, 0, &linput_mux),
 SND_SOC_DAPM_MUX("Left Input Inverting Mux", SND_SOC_NOPM, 0, 0,
@@ -947,7 +947,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0),
 static const struct snd_soc_dapm_route wm8903_intercon[] = {
 
 	{ "CLK_DSP", NULL, "CLK_SYS" },
-	{ "Mic Bias", NULL, "CLK_SYS" },
+	{ "MICBIAS", NULL, "CLK_SYS" },
 	{ "HPL_DCS", NULL, "CLK_SYS" },
 	{ "HPR_DCS", NULL, "CLK_SYS" },
 	{ "LINEOUTL_DCS", NULL, "CLK_SYS" },
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index b260f54a4462..2f5b1074a8d9 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -201,8 +201,8 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = {
 	{"Int Spk", NULL, "RON"},
 	{"Int Spk", NULL, "LOP"},
 	{"Int Spk", NULL, "LON"},
-	{"Mic Bias", NULL, "Mic Jack"},
-	{"IN1L", NULL, "Mic Bias"},
+	{"Mic Jack", NULL, "MICBIAS"},
+	{"IN1L", NULL, "Mic Jack"},
 };
 
 static const struct snd_soc_dapm_route seaboard_audio_map[] = {
@@ -212,8 +212,8 @@ static const struct snd_soc_dapm_route seaboard_audio_map[] = {
 	{"Int Spk", NULL, "RON"},
 	{"Int Spk", NULL, "LOP"},
 	{"Int Spk", NULL, "LON"},
-	{"Mic Bias", NULL, "Mic Jack"},
-	{"IN1R", NULL, "Mic Bias"},
+	{"Mic Jack", NULL, "MICBIAS"},
+	{"IN1R", NULL, "Mic Jack"},
 };
 
 static const struct snd_soc_dapm_route kaen_audio_map[] = {
@@ -223,8 +223,8 @@ static const struct snd_soc_dapm_route kaen_audio_map[] = {
 	{"Int Spk", NULL, "RON"},
 	{"Int Spk", NULL, "LOP"},
 	{"Int Spk", NULL, "LON"},
-	{"Mic Bias", NULL, "Mic Jack"},
-	{"IN2R", NULL, "Mic Bias"},
+	{"Mic Jack", NULL, "MICBIAS"},
+	{"IN2R", NULL, "Mic Jack"},
 };
 
 static const struct snd_soc_dapm_route aebl_audio_map[] = {
@@ -232,8 +232,8 @@ static const struct snd_soc_dapm_route aebl_audio_map[] = {
 	{"Headphone Jack", NULL, "HPOUTL"},
 	{"Int Spk", NULL, "LINEOUTR"},
 	{"Int Spk", NULL, "LINEOUTL"},
-	{"Mic Bias", NULL, "Mic Jack"},
-	{"IN1R", NULL, "Mic Bias"},
+	{"Mic Jack", NULL, "MICBIAS"},
+	{"IN1R", NULL, "Mic Jack"},
 };
 
 static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
@@ -329,7 +329,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE,
 				0);
 
-	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+	snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
 
 	return 0;
 }
-- 
cgit v1.2.3


From af3c2621a9b4d22b8927b91bc9cc02a13087e12b Mon Sep 17 00:00:00 2001
From: Axel Lin <axel.lin@gmail.com>
Date: Mon, 28 Nov 2011 18:55:03 +0800
Subject: ASoC: Convert tegra_spdif to use module_platform_driver()

Use the module_platform_driver() macro which makes
the code smaller and a bit simpler.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_spdif.c | 12 +-----------
 1 file changed, 1 insertion(+), 11 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
index ea9c92036aa1..475428cf270e 100644
--- a/sound/soc/tegra/tegra_spdif.c
+++ b/sound/soc/tegra/tegra_spdif.c
@@ -352,17 +352,7 @@ static struct platform_driver tegra_spdif_driver = {
 	.remove = __devexit_p(tegra_spdif_platform_remove),
 };
 
-static int __init snd_tegra_spdif_init(void)
-{
-	return platform_driver_register(&tegra_spdif_driver);
-}
-module_init(snd_tegra_spdif_init);
-
-static void __exit snd_tegra_spdif_exit(void)
-{
-	platform_driver_unregister(&tegra_spdif_driver);
-}
-module_exit(snd_tegra_spdif_exit);
+module_platform_driver(tegra_spdif_driver);
 
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra SPDIF ASoC driver");
-- 
cgit v1.2.3


From bf55499e6ee927e047feed85349365481289bd75 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 29 Nov 2011 18:36:48 -0700
Subject: ASoC: Tegra I2S: Add device tree binding

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 .../devicetree/bindings/sound/tegra20-i2s.txt      | 17 ++++++++++++++
 sound/soc/tegra/tegra_i2s.c                        | 27 ++++++++++++++++++----
 2 files changed, 39 insertions(+), 5 deletions(-)
 create mode 100644 Documentation/devicetree/bindings/sound/tegra20-i2s.txt

(limited to 'sound/soc/tegra')

diff --git a/Documentation/devicetree/bindings/sound/tegra20-i2s.txt b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt
new file mode 100644
index 000000000000..0df2b5c816e3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt
@@ -0,0 +1,17 @@
+NVIDIA Tegra 20 I2S controller
+
+Required properties:
+- compatible : "nvidia,tegra20-i2s"
+- reg : Should contain I2S registers location and length
+- interrupts : Should contain I2S interrupt
+- nvidia,dma-request-selector : The Tegra DMA controller's phandle and
+  request selector for this I2S controller
+
+Example:
+
+i2s@70002800 {
+	compatible = "nvidia,tegra20-i2s";
+	reg = <0x70002800 0x200>;
+	interrupts = < 45 >;
+	nvidia,dma-request-selector = < &apbdma 2 >;
+};
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index ca4d0c0a913e..33509de52540 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -36,6 +36,7 @@
 #include <linux/seq_file.h>
 #include <linux/slab.h>
 #include <linux/io.h>
+#include <linux/of.h>
 #include <mach/iomap.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -331,6 +332,8 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 {
 	struct tegra_i2s * i2s;
 	struct resource *mem, *memregion, *dmareq;
+	u32 of_dma[2];
+	u32 dma_ch;
 	int ret;
 
 	i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL);
@@ -360,9 +363,16 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 
 	dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!dmareq) {
-		dev_err(&pdev->dev, "No DMA resource\n");
-		ret = -ENODEV;
-		goto err_clk_put;
+		if (of_property_read_u32_array(pdev->dev.of_node,
+					"nvidia,dma-request-selector",
+					of_dma, 2) < 0) {
+			dev_err(&pdev->dev, "No DMA resource\n");
+			ret = -ENODEV;
+			goto err_clk_put;
+		}
+		dma_ch = of_dma[1];
+	} else {
+		dma_ch = dmareq->start;
 	}
 
 	memregion = devm_request_mem_region(&pdev->dev, mem->start,
@@ -383,12 +393,12 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 	i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2;
 	i2s->capture_dma_data.wrap = 4;
 	i2s->capture_dma_data.width = 32;
-	i2s->capture_dma_data.req_sel = dmareq->start;
+	i2s->capture_dma_data.req_sel = dma_ch;
 
 	i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1;
 	i2s->playback_dma_data.wrap = 4;
 	i2s->playback_dma_data.width = 32;
-	i2s->playback_dma_data.req_sel = dmareq->start;
+	i2s->playback_dma_data.req_sel = dma_ch;
 
 	i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED;
 
@@ -422,10 +432,16 @@ static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct of_device_id tegra_i2s_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra20-i2s", },
+	{},
+};
+
 static struct platform_driver tegra_i2s_driver = {
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.of_match_table = tegra_i2s_of_match,
 	},
 	.probe = tegra_i2s_platform_probe,
 	.remove = __devexit_p(tegra_i2s_platform_remove),
@@ -436,3 +452,4 @@ MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra I2S ASoC driver");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra_i2s_of_match);
-- 
cgit v1.2.3


From 7b9b5e11704afb8f05aa6490c3b4bb2cc328647c Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 7 Dec 2011 13:58:29 -0700
Subject: ASoC: Tegra: Move DAS configuration into DAS driver

Move DAS routing setup into the DAS driver itself. This removes the need
to duplicate this in each machine driver, of which we'll soon have three.

An added advantage is that the machine drivers no longer call the Tegra20-
specific DAS functions by name, so the machine driver no longer needs to
be split up into Tegra20 and Tegra30 versions.

If individual machine drivers need a different routing setup to this
default, they can still call the DAS functions to set that up.

Long-term, DAS will be a codec driver, and user-space will be able to
control its routing, possibly within constraints that the machine driver
sets up. Configuring the DAS routing from the DAS driver is a very slight
move in that direction.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_das.c    | 13 +++++++++++++
 sound/soc/tegra/tegra_wm8903.c | 13 -------------
 sound/soc/tegra/trimslice.c    | 23 -----------------------
 3 files changed, 13 insertions(+), 36 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c
index 5b82b4e79231..3b3c1ba4d235 100644
--- a/sound/soc/tegra/tegra_das.c
+++ b/sound/soc/tegra/tegra_das.c
@@ -202,6 +202,19 @@ static int __devinit tegra_das_probe(struct platform_device *pdev)
 		goto err;
 	}
 
+	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1,
+					   TEGRA_DAS_DAP_SEL_DAC1);
+	if (ret) {
+		dev_err(&pdev->dev, "Can't set up DAS DAP connection\n");
+		goto err;
+	}
+	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1,
+					   TEGRA_DAS_DAC_SEL_DAP1);
+	if (ret) {
+		dev_err(&pdev->dev, "Can't set up DAS DAC connection\n");
+		goto err;
+	}
+
 	tegra_das_debug_add(das);
 
 	platform_set_drvdata(pdev, das);
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 2f5b1074a8d9..ba2d23ea6424 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -249,19 +249,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	struct tegra_wm8903_platform_data *pdata = machine->pdata;
 	int ret;
 
-	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1,
-					   TEGRA_DAS_DAP_SEL_DAC1);
-	if (ret) {
-		dev_err(card->dev, "Can't set up DAS DAP connection\n");
-		return ret;
-	}
-	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1,
-					   TEGRA_DAS_DAC_SEL_DAP1);
-	if (ret) {
-		dev_err(card->dev, "Can't set up DAS DAC connection\n");
-		return ret;
-	}
-
 	if (gpio_is_valid(pdata->gpio_spkr_en)) {
 		ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
 		if (ret) {
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 043eb7c7eb73..7d95b7697a73 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -115,28 +115,6 @@ static const struct snd_soc_dapm_route trimslice_audio_map[] = {
 	{"RLINEIN", NULL, "Line In"},
 };
 
-static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_card *card = codec->card;
-	int ret;
-
-	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1,
-					   TEGRA_DAS_DAP_SEL_DAC1);
-	if (ret) {
-		dev_err(card->dev, "Can't set up DAS DAP connection\n");
-		return ret;
-	}
-	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1,
-					   TEGRA_DAS_DAC_SEL_DAP1);
-	if (ret) {
-		dev_err(card->dev, "Can't set up DAS DAC connection\n");
-		return ret;
-	}
-
-	return 0;
-}
-
 static struct snd_soc_dai_link trimslice_tlv320aic23_dai = {
 	.name = "TLV320AIC23",
 	.stream_name = "AIC23",
@@ -144,7 +122,6 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = {
 	.platform_name = "tegra-pcm-audio",
 	.cpu_dai_name = "tegra-i2s.0",
 	.codec_dai_name = "tlv320aic23-hifi",
-	.init = trimslice_asoc_init,
 	.ops = &trimslice_asoc_ops,
 };
 
-- 
cgit v1.2.3


From 58783faf281559379871d85faf2ef53e97d075e0 Mon Sep 17 00:00:00 2001
From: Leon Romanovsky <leon@leon.nu>
Date: Mon, 19 Dec 2011 21:51:52 +0200
Subject: ASoC: Tegra machine ASoC driver for boards using ALC5332 codec

At this stage only Toshiba AC100/Dynabook supported.

Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/Kconfig         |   9 ++
 sound/soc/tegra/Makefile        |   2 +
 sound/soc/tegra/tegra_alc5632.c | 213 ++++++++++++++++++++++++++++++++++++++++
 3 files changed, 224 insertions(+)
 create mode 100644 sound/soc/tegra/tegra_alc5632.c

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index c6af1fd707f5..ce1b773c351f 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE
 	help
 	  Say Y or M here if you want to add support for SoC audio on the
 	  TrimSlice platform.
+
+config SND_SOC_TEGRA_ALC5632
+       tristate "SoC Audio support for Tegra boards using an ALC5632 codec"
+       depends on SND_SOC_TEGRA && I2C
+       select SND_SOC_TEGRA_I2S
+       select SND_SOC_ALC5632
+       help
+         Say Y or M here if you want to add support for SoC audio on the
+         Toshiba AC100 netbook.
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index 4d943b3fe150..8e584b8fcfba 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o
 # Tegra machine Support
 snd-soc-tegra-wm8903-objs := tegra_wm8903.o
 snd-soc-tegra-trimslice-objs := trimslice.o
+snd-soc-tegra-alc5632-objs := tegra_alc5632.o
 
 obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o
 obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o
+obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
new file mode 100644
index 000000000000..9287eb8028fd
--- /dev/null
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -0,0 +1,213 @@
+/*
+* tegra_alc5632.c  --  Toshiba AC100(PAZ00) machine ASoC driver
+*
+* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
+*
+* Authors:  Leon Romanovsky <leon@leon.nu>
+*           Andrey Danin <danindrey@mail.ru>
+*           Marc Dietrich <marvin24@gmx.de>
+*
+* 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 <asm/mach-types.h>
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "../codecs/alc5632.h"
+
+#include "tegra_das.h"
+#include "tegra_i2s.h"
+#include "tegra_pcm.h"
+#include "tegra_asoc_utils.h"
+
+#define DRV_NAME "tegra-alc5632"
+
+struct tegra_alc5632 {
+	struct tegra_asoc_utils_data util_data;
+};
+
+static int tegra_alc5632_asoc_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;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = codec->card;
+	struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
+	int srate, mclk;
+	int err;
+
+	srate = params_rate(params);
+	mclk = 512 * srate;
+
+	err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk);
+	if (err < 0) {
+		dev_err(card->dev, "Can't configure clocks\n");
+		return err;
+	}
+
+	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+					SND_SOC_CLOCK_IN);
+	if (err < 0) {
+		dev_err(card->dev, "codec_dai clock not set\n");
+		return err;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops tegra_alc5632_asoc_ops = {
+	.hw_params = tegra_alc5632_asoc_hw_params,
+};
+
+static struct snd_soc_jack tegra_alc5632_hs_jack;
+
+static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Int Spk", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = {
+	/* Internal Speaker */
+	{"Int Spk", NULL, "SPKOUT"},
+	{"Int Spk", NULL, "SPKOUTN"},
+
+	/* Headset Mic */
+	{"MIC1", NULL, "MICBIAS1"},
+	{"MICBIAS1", NULL, "Headset Mic"},
+
+	/* Headset Stereophone */
+	{"Headset Stereophone", NULL, "HPR"},
+	{"Headset Stereophone", NULL, "HPL"},
+};
+
+static const struct snd_kcontrol_new tegra_alc5632_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Int Spk"),
+};
+
+static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
+			 &tegra_alc5632_hs_jack);
+	snd_soc_jack_add_pins(&tegra_alc5632_hs_jack,
+			ARRAY_SIZE(tegra_alc5632_hs_jack_pins),
+			tegra_alc5632_hs_jack_pins);
+
+	snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1");
+
+	return 0;
+}
+
+static struct snd_soc_dai_link tegra_alc5632_dai = {
+	.name = "ALC5632",
+	.stream_name = "ALC5632 PCM",
+	.codec_name = "alc5632.0-001e",
+	.platform_name = "tegra-pcm-audio",
+	.cpu_dai_name = "tegra-i2s.0",
+	.codec_dai_name = "alc5632-hifi",
+	.init = tegra_alc5632_asoc_init,
+	.ops = &tegra_alc5632_asoc_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S
+			   | SND_SOC_DAIFMT_NB_NF
+			   | SND_SOC_DAIFMT_CBS_CFS,
+};
+
+static struct snd_soc_card snd_soc_tegra_alc5632 = {
+	.name = "tegra-alc5632",
+	.dai_link = &tegra_alc5632_dai,
+	.num_links = 1,
+	.controls = tegra_alc5632_controls,
+	.num_controls = ARRAY_SIZE(tegra_alc5632_controls),
+	.dapm_widgets = tegra_alc5632_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets),
+	.dapm_routes = tegra_alc5632_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map),
+	.fully_routed = true,
+};
+
+static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &snd_soc_tegra_alc5632;
+	struct tegra_alc5632 *alc5632;
+	int ret;
+
+	alc5632 = devm_kzalloc(&pdev->dev,
+			sizeof(struct tegra_alc5632), GFP_KERNEL);
+	if (!alc5632) {
+		dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n");
+		return -ENOMEM;
+	}
+
+	ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
+	if (ret)
+		return ret;
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, alc5632);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+			ret);
+		tegra_asoc_utils_fini(&alc5632->util_data);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit tegra_alc5632_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
+
+	snd_soc_unregister_card(card);
+
+	tegra_asoc_utils_fini(&alc5632->util_data);
+
+	return 0;
+}
+
+static struct platform_driver tegra_alc5632_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = tegra_alc5632_probe,
+	.remove = __devexit_p(tegra_alc5632_remove),
+};
+module_platform_driver(tegra_alc5632_driver);
+
+MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>");
+MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
-- 
cgit v1.2.3


From 07cdf36d8c4ba4ad0db13228eb25bcd3d5138b29 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Mon, 12 Dec 2011 15:55:36 -0700
Subject: ASoC: Tegra+WM8903 machine: Add device tree binding

This driver is parameterized in two ways:

a) Platform data, which supplies the set of GPIOs used by the driver.
   These GPIOs can now be parsed out of device tree.

b) Machine-specific DAPM route arrays embedded into the ASoC machine
   driver itself. Historically, the driver picks the appropriate array
   to use using machine_is_*(). The driver now requires this array to
   be parsed from device tree when instantiated through device tree,
   using the core ASoC support for this parsing.

Based on work by John Bonesio, but significantly reworked since then.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 .../bindings/sound/tegra-audio-wm8903.txt          |  71 ++++++++++++
 sound/soc/tegra/tegra_wm8903.c                     | 128 +++++++++++++++++----
 2 files changed, 174 insertions(+), 25 deletions(-)
 create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt

(limited to 'sound/soc/tegra')

diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt
new file mode 100644
index 000000000000..d5b0da8bf1d8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt
@@ -0,0 +1,71 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm8903"
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : A list of the connections between audio components.
+  Each entry is a pair of strings, the first being the connection's sink,
+  the second being the connection's source. Valid names for sources and
+  sinks are the WM8903's pins, and the jacks on the board:
+
+  WM8903 pins:
+
+  * IN1L
+  * IN1R
+  * IN2L
+  * IN2R
+  * IN3L
+  * IN3R
+  * DMICDAT
+  * HPOUTL
+  * HPOUTR
+  * LINEOUTL
+  * LINEOUTR
+  * LOP
+  * LON
+  * ROP
+  * RON
+  * MICBIAS
+
+  Board connectors:
+
+  * Headphone Jack
+  * Int Spk
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8903 audio codec
+
+Optional properties:
+- nvidia,spkr-en-gpios : The GPIO that enables the speakers
+- nvidia,hp-mute-gpios : The GPIO that mutes the headphones
+- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone
+- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-wm8903-harmony",
+		     "nvidia,tegra-audio-wm8903"
+	nvidia,model = "tegra-wm8903-harmony";
+
+	nvidia,audio-routing =
+		"Headphone Jack", "HPOUTR",
+		"Headphone Jack", "HPOUTL",
+		"Int Spk", "ROP",
+		"Int Spk", "RON",
+		"Int Spk", "LOP",
+		"Int Spk", "LON",
+		"Mic Jack", "MICBIAS",
+		"IN1L", "Mic Jack";
+
+	nvidia,i2s-controller = <&i2s1>;
+	nvidia,audio-codec = <&wm8903>;
+
+	nvidia,spkr-en-gpios = <&codec 2 0>;
+	nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */
+	nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */
+	nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */
+};
+
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index ba2d23ea6424..4677f2666300 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -34,6 +34,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/gpio.h>
+#include <linux/of_gpio.h>
 
 #include <mach/tegra_wm8903_pdata.h>
 
@@ -59,8 +60,9 @@
 #define GPIO_HP_DET     BIT(4)
 
 struct tegra_wm8903 {
+	struct tegra_wm8903_platform_data pdata;
+	struct platform_device *pcm_dev;
 	struct tegra_asoc_utils_data util_data;
-	struct tegra_wm8903_platform_data *pdata;
 	int gpio_requested;
 };
 
@@ -160,7 +162,7 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w,
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_card *card = dapm->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = machine->pdata;
+	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
 	if (!(machine->gpio_requested & GPIO_SPKR_EN))
 		return 0;
@@ -177,7 +179,7 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w,
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_card *card = dapm->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = machine->pdata;
+	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
 	if (!(machine->gpio_requested & GPIO_HP_MUTE))
 		return 0;
@@ -246,9 +248,36 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = machine->pdata;
+	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
+	struct device_node *np = card->dev->of_node;
 	int ret;
 
+	if (card->dev->platform_data) {
+		memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
+	} else if (np) {
+		/*
+		 * This part must be in init() rather than probe() in order to
+		 * guarantee that the WM8903 has been probed, and hence its
+		 * GPIO controller registered, which is a pre-condition for
+		 * of_get_named_gpio() to be able to map the phandles in the
+		 * properties to the controller node. Given this, all
+		 * pdata handling is in init() for consistency.
+		 */
+		pdata->gpio_spkr_en = of_get_named_gpio(np,
+						"nvidia,spkr-en-gpios", 0);
+		pdata->gpio_hp_mute = of_get_named_gpio(np,
+						"nvidia,hp-mute-gpios", 0);
+		pdata->gpio_hp_det = of_get_named_gpio(np,
+						"nvidia,hp-det-gpios", 0);
+		pdata->gpio_int_mic_en = of_get_named_gpio(np,
+						"nvidia,int-mic-en-gpios", 0);
+		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
+						"nvidia,ext-mic-en-gpios", 0);
+	} else {
+		dev_err(card->dev, "No platform data supplied\n");
+		return -EINVAL;
+	}
+
 	if (gpio_is_valid(pdata->gpio_spkr_en)) {
 		ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
 		if (ret) {
@@ -348,11 +377,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &snd_soc_tegra_wm8903;
 	struct tegra_wm8903 *machine;
-	struct tegra_wm8903_platform_data *pdata;
 	int ret;
 
-	pdata = pdev->dev.platform_data;
-	if (!pdata) {
+	if (!pdev->dev.platform_data && !pdev->dev.of_node) {
 		dev_err(&pdev->dev, "No platform data supplied\n");
 		return -EINVAL;
 	}
@@ -364,31 +391,70 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		ret = -ENOMEM;
 		goto err;
 	}
-
-	machine->pdata = pdata;
-
-	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
-	if (ret)
-		goto err;
+	machine->pcm_dev = ERR_PTR(-EINVAL);
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
-	if (machine_is_harmony()) {
-		card->dapm_routes = harmony_audio_map;
-		card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
-	} else if (machine_is_seaboard()) {
-		card->dapm_routes = seaboard_audio_map;
-		card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map);
-	} else if (machine_is_kaen()) {
-		card->dapm_routes = kaen_audio_map;
-		card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map);
+	if (pdev->dev.of_node) {
+		ret = snd_soc_of_parse_card_name(card, "nvidia,model");
+		if (ret)
+			goto err;
+
+		ret = snd_soc_of_parse_audio_routing(card,
+						     "nvidia,audio-routing");
+		if (ret)
+			goto err;
+
+		tegra_wm8903_dai.codec_name = NULL;
+		tegra_wm8903_dai.codec_of_node = of_parse_phandle(
+				pdev->dev.of_node, "nvidia,audio-codec", 0);
+		if (!tegra_wm8903_dai.codec_of_node) {
+			dev_err(&pdev->dev,
+				"Property 'nvidia,audio-codec' missing or invalid\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		tegra_wm8903_dai.cpu_dai_name = NULL;
+		tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(
+				pdev->dev.of_node, "nvidia,i2s-controller", 0);
+		if (!tegra_wm8903_dai.cpu_dai_of_node) {
+			dev_err(&pdev->dev,
+				"Property 'nvidia,i2s-controller' missing or invalid\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		machine->pcm_dev = platform_device_register_simple(
+					"tegra-pcm-audio", -1, NULL, 0);
+		if (IS_ERR(machine->pcm_dev)) {
+			dev_err(&pdev->dev,
+				"Can't instantiate tegra-pcm-audio\n");
+			ret = PTR_ERR(machine->pcm_dev);
+			goto err;
+		}
 	} else {
-		card->dapm_routes = aebl_audio_map;
-		card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map);
+		if (machine_is_harmony()) {
+			card->dapm_routes = harmony_audio_map;
+			card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
+		} else if (machine_is_seaboard()) {
+			card->dapm_routes = seaboard_audio_map;
+			card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map);
+		} else if (machine_is_kaen()) {
+			card->dapm_routes = kaen_audio_map;
+			card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map);
+		} else {
+			card->dapm_routes = aebl_audio_map;
+			card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map);
+		}
 	}
 
+	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
+	if (ret)
+		goto err_unregister;
+
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
@@ -400,6 +466,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
+err_unregister:
+	if (!IS_ERR(machine->pcm_dev))
+		platform_device_unregister(machine->pcm_dev);
 err:
 	return ret;
 }
@@ -408,7 +477,7 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = machine->pdata;
+	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
 	if (machine->gpio_requested & GPIO_HP_DET)
 		snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
@@ -427,15 +496,23 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 	snd_soc_unregister_card(card);
 
 	tegra_asoc_utils_fini(&machine->util_data);
+	if (!IS_ERR(machine->pcm_dev))
+		platform_device_unregister(machine->pcm_dev);
 
 	return 0;
 }
 
+static const struct of_device_id tegra_wm8903_of_match[] __devinitconst = {
+	{ .compatible = "nvidia,tegra-audio-wm8903", },
+	{},
+};
+
 static struct platform_driver tegra_wm8903_driver = {
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
 		.pm = &snd_soc_pm_ops,
+		.of_match_table = tegra_wm8903_of_match,
 	},
 	.probe = tegra_wm8903_driver_probe,
 	.remove = __devexit_p(tegra_wm8903_driver_remove),
@@ -446,3 +523,4 @@ MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra+WM8903 machine ASoC driver");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra_wm8903_of_match);
-- 
cgit v1.2.3


From b16eaf9fd324a70ecca48faa7ef3f349baf7f0cd Mon Sep 17 00:00:00 2001
From: Axel Lin <axel.lin@gmail.com>
Date: Thu, 22 Dec 2011 21:23:01 +0800
Subject: ASoC: tegra: Add .owner to struct snd_soc_card
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.

Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_alc5632.c | 1 +
 sound/soc/tegra/tegra_wm8903.c  | 1 +
 sound/soc/tegra/trimslice.c     | 1 +
 3 files changed, 3 insertions(+)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 9287eb8028fd..4a0e805c4edd 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -141,6 +141,7 @@ static struct snd_soc_dai_link tegra_alc5632_dai = {
 
 static struct snd_soc_card snd_soc_tegra_alc5632 = {
 	.name = "tegra-alc5632",
+	.owner = THIS_MODULE,
 	.dai_link = &tegra_alc5632_dai,
 	.num_links = 1,
 	.controls = tegra_alc5632_controls,
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 4677f2666300..566655e23b7d 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -363,6 +363,7 @@ static struct snd_soc_dai_link tegra_wm8903_dai = {
 
 static struct snd_soc_card snd_soc_tegra_wm8903 = {
 	.name = "tegra-wm8903",
+	.owner = THIS_MODULE,
 	.dai_link = &tegra_wm8903_dai,
 	.num_links = 1,
 
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 7d95b7697a73..2bdfc550cff8 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -127,6 +127,7 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = {
 
 static struct snd_soc_card snd_soc_trimslice = {
 	.name = "tegra-trimslice",
+	.owner = THIS_MODULE,
 	.dai_link = &trimslice_tlv320aic23_dai,
 	.num_links = 1,
 
-- 
cgit v1.2.3


From 25e9e7565f9aa9e4b976387a3fab60bfaa4efac8 Mon Sep 17 00:00:00 2001
From: Joachim Eastwood <joachim.eastwood@jotron.com>
Date: Sun, 1 Jan 2012 01:58:44 +0100
Subject: ASoC: check for substream not channels_min in pcm engines

This is a follow up on 53dea36c70c1857 which fixes the other affected
pcm engines.

Description from 53dea36c70c1857:
 Don't rely on the codec's channels_min information to decide wheter or
 not allocate a substream's DMA buffer. Rather check if the substream
 itself was allocated previously.

Without this patch I was seeing null-pointer dereferenc in atmel-pcm.

Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel-pcm.c         | 5 ++---
 sound/soc/blackfin/bf5xx-ac97-pcm.c | 5 ++---
 sound/soc/blackfin/bf5xx-i2s-pcm.c  | 5 ++---
 sound/soc/blackfin/bf5xx-tdm-pcm.c  | 5 ++---
 sound/soc/davinci/davinci-pcm.c     | 5 ++---
 sound/soc/ep93xx/ep93xx-pcm.c       | 5 ++---
 sound/soc/jz4740/jz4740-pcm.c       | 5 ++---
 sound/soc/kirkwood/kirkwood-dma.c   | 5 ++---
 sound/soc/mid-x86/sst_platform.c    | 5 ++---
 sound/soc/omap/omap-pcm.c           | 5 ++---
 sound/soc/samsung/dma.c             | 5 ++---
 sound/soc/samsung/idma.c            | 3 +--
 sound/soc/tegra/tegra_pcm.c         | 5 ++---
 13 files changed, 25 insertions(+), 38 deletions(-)

(limited to 'sound/soc/tegra')

diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 60de05525c06..a21ff459e5d3 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -367,7 +367,6 @@ static u64 atmel_pcm_dmamask = 0xffffffff;
 static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -376,14 +375,14 @@ static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = atmel_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		pr_debug("atmel-pcm:"
 				"Allocating PCM capture DMA buffer\n");
 		ret = atmel_pcm_preallocate_dma_buffer(pcm,
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index fcff58390848..d7dc9bde0976 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -421,7 +421,6 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -431,14 +430,14 @@ static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 6ec3d41b9b6d..63205d723eab 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -260,7 +260,6 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -270,14 +269,14 @@ static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c
index 4406f9a865ae..254490cf1876 100644
--- a/sound/soc/blackfin/bf5xx-tdm-pcm.c
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c
@@ -286,7 +286,6 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -295,14 +294,14 @@ static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 65bff3d30dd7..b26401f87b85 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -831,7 +831,6 @@ static u64 davinci_pcm_dmamask = 0xffffffff;
 static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
@@ -840,7 +839,7 @@ static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = davinci_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK,
 			pcm_hardware_playback.buffer_bytes_max);
@@ -848,7 +847,7 @@ static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd)
 			return ret;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = davinci_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE,
 			pcm_hardware_capture.buffer_bytes_max);
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index a2de9c42b702..3fc96130d1a6 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -286,7 +286,6 @@ static u64 ep93xx_pcm_dmamask = 0xffffffff;
 static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -295,14 +294,14 @@ static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = ep93xx_pcm_preallocate_dma_buffer(pcm,
 					SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			return ret;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = ep93xx_pcm_preallocate_dma_buffer(pcm,
 					SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c
index 50cda9ea9156..9b8cf256847d 100644
--- a/sound/soc/jz4740/jz4740-pcm.c
+++ b/sound/soc/jz4740/jz4740-pcm.c
@@ -302,7 +302,6 @@ static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32);
 static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -312,14 +311,14 @@ static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = jz4740_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto err;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = jz4740_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index 210438261a49..d4a17780cef4 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -315,7 +315,6 @@ static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm,
 static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
@@ -324,14 +323,14 @@ static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = kirkwood_dma_preallocate_dma_buffer(pcm,
 				SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			return ret;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = kirkwood_dma_preallocate_dma_buffer(pcm,
 				SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index c2bf172a196e..d34563b12c3b 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -446,13 +446,12 @@ static void sst_pcm_free(struct snd_pcm *pcm)
 
 static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int retval = 0;
 
 	pr_debug("sst_pcm_new called\n");
-	if (dai->driver->playback.channels_min ||
-			dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
+			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		retval =  snd_pcm_lib_preallocate_pages_for_all(pcm,
 			SNDRV_DMA_TYPE_CONTINUOUS,
 			snd_dma_continuous_data(GFP_KERNEL),
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 52a0f634948e..a59bd352d342 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -378,7 +378,6 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
 static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -387,14 +386,14 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(64);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = omap_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = omap_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 797c3d5e79e5..427ae0d9817b 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -403,7 +403,6 @@ static u64 dma_mask = DMA_BIT_MASK(32);
 static int dma_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -414,14 +413,14 @@ static int dma_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index 2bcf75815624..3ba6aba8e2b9 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -387,7 +387,6 @@ static u64 idma_mask = DMA_BIT_MASK(32);
 static int idma_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -396,7 +395,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min)
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = preallocate_idma_buffer(pcm,
 				SNDRV_PCM_STREAM_PLAYBACK);
 
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 90345ee138f3..c22431516ab2 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -330,7 +330,6 @@ static u64 tegra_dma_mask = DMA_BIT_MASK(32);
 static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
@@ -339,14 +338,14 @@ static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = tegra_pcm_preallocate_dma_buffer(pcm,
 						SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto err;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = tegra_pcm_preallocate_dma_buffer(pcm,
 						SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
-- 
cgit v1.2.3