From: Eric Miao <eric.y.miao@gmail.com>
Date: Tue, 3 Mar 2009 01:41:00 +0000 (+0800)
Subject: ASoC: make ops a pointer in 'struct snd_soc_dai'
X-Git-Tag: MMI-PSA29.97-13-9~28576^2~33^2
X-Git-Url: https://git.stricted.de/?a=commitdiff_plain;h=6335d05548eece40092000aa91b64a50310d69d5;p=GitHub%2FMotorolaMobilityLLC%2Fkernel-slsi.git

ASoC: make ops a pointer in 'struct snd_soc_dai'

Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.

The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.

Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 24247f763608..13676472ddfc 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -203,7 +203,7 @@ struct snd_soc_dai {
 	int (*resume)(struct snd_soc_dai *dai);
 
 	/* ops */
-	struct snd_soc_dai_ops ops;
+	struct snd_soc_dai_ops *ops;
 
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index ff0054b76502..e588e63f18d2 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_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 = {
+	.startup	= atmel_ssc_startup,
+	.shutdown	= atmel_ssc_shutdown,
+	.prepare	= atmel_ssc_prepare,
+	.hw_params	= atmel_ssc_hw_params,
+	.set_fmt	= atmel_ssc_set_dai_fmt,
+	.set_clkdiv	= atmel_ssc_set_dai_clkdiv,
+};
+
 struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 	{	.name = "atmel-ssc0",
 		.id = 0,
@@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[0],
 	},
 #if NUM_SSC_DEVICES == 3
@@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[1],
 	},
 	{	.name = "atmel-ssc2",
@@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[2],
 	},
 #endif
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index f0e30aec7f23..479d7bdf1865 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.trigger	= au1xpsc_ac97_trigger,
+	.hw_params	= au1xpsc_ac97_hw_params,
+};
+
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
 	.ac97_control		= 1,
@@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.trigger	= au1xpsc_ac97_trigger,
-		.hw_params	= au1xpsc_ac97_hw_params,
-	},
+	.ops = &au1xpsc_ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index f916de4400ed..bb589327ee32 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.trigger	= au1xpsc_i2s_trigger,
+	.hw_params	= au1xpsc_i2s_hw_params,
+	.set_fmt	= au1xpsc_i2s_set_fmt,
+};
+
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
 	.probe			= au1xpsc_i2s_probe,
@@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 		.channels_min	= 2,
 		.channels_max	= 8,	/* 2 without external help */
 	},
-	.ops = {
-		.trigger	= au1xpsc_i2s_trigger,
-		.hw_params	= au1xpsc_i2s_hw_params,
-		.set_fmt	= au1xpsc_i2s_set_fmt,
-	},
+	.ops = &au1xpsc_i2s_dai_ops,
 };
 EXPORT_SYMBOL(au1xpsc_i2s_dai);
 
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index d1d95d2393fe..964824419678 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev,
 #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\
 	SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
+	.startup	= bf5xx_i2s_startup,
+	.shutdown	= bf5xx_i2s_shutdown,
+	.hw_params	= bf5xx_i2s_hw_params,
+	.set_fmt	= bf5xx_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai bf5xx_i2s_dai = {
 	.name = "bf5xx-i2s",
 	.id = 0,
@@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
-	.ops = {
-		.startup   = bf5xx_i2s_startup,
-		.shutdown  = bf5xx_i2s_shutdown,
-		.hw_params = bf5xx_i2s_hw_params,
-		.set_fmt = bf5xx_i2s_set_dai_fmt,
-	},
+	.ops = &bf5xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 11f84b6e5cb8..b0d4af145b87 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -41,6 +41,10 @@ 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 = {
+	.prepare	= ac97_prepare,
+};
+
 struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
 	.ac97_control = 1,
@@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = {
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ac97_dai);
 
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index d56e6bb1fedb..1f63d387a2f4 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -421,6 +421,13 @@ 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 = {
+	.hw_params	= ak4535_hw_params,
+	.set_fmt	= ak4535_set_dai_fmt,
+	.digital_mute	= ak4535_mute,
+	.set_sysclk	= ak4535_set_dai_sysclk,
+};
+
 struct snd_soc_dai ak4535_dai = {
 	.name = "AK4535",
 	.playback = {
@@ -435,12 +442,7 @@ struct snd_soc_dai ak4535_dai = {
 		.channels_max = 2,
 		.rates = AK4535_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = ak4535_hw_params,
-		.set_fmt = ak4535_set_dai_fmt,
-		.digital_mute = ak4535_mute,
-		.set_sysclk = ak4535_set_dai_sysclk,
-	},
+	.ops = &ak4535_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ak4535_dai);
 
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f86f33cc1798..7ae3d6520e3f 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -503,6 +503,13 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
  */
 static struct snd_soc_codec *cs4270_codec;
 
+static 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,
+	.digital_mute	= cs4270_mute,
+};
+
 struct snd_soc_dai cs4270_dai = {
 	.name = "cs4270",
 	.playback = {
@@ -519,12 +526,7 @@ struct snd_soc_dai cs4270_dai = {
 		.rates = 0,
 		.formats = CS4270_FORMATS,
 	},
-	.ops = {
-		.hw_params = cs4270_hw_params,
-		.set_sysclk = cs4270_set_dai_sysclk,
-		.set_fmt = cs4270_set_dai_fmt,
-		.digital_mute = cs4270_mute,
-	},
+	.ops = &cs4270_dai_ops,
 };
 EXPORT_SYMBOL_GPL(cs4270_dai);
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 58e225dadc7e..87f606c76822 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -506,6 +506,16 @@ 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 = {
+	.startup	= ssm2602_startup,
+	.prepare	= ssm2602_pcm_prepare,
+	.hw_params	= ssm2602_hw_params,
+	.shutdown	= ssm2602_shutdown,
+	.digital_mute	= ssm2602_mute,
+	.set_sysclk	= ssm2602_set_dai_sysclk,
+	.set_fmt	= ssm2602_set_dai_fmt,
+};
+
 struct snd_soc_dai ssm2602_dai = {
 	.name = "SSM2602",
 	.playback = {
@@ -520,15 +530,7 @@ struct snd_soc_dai ssm2602_dai = {
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
 		.formats = SSM2602_FORMATS,},
-	.ops = {
-		.startup = ssm2602_startup,
-		.prepare = ssm2602_pcm_prepare,
-		.hw_params = ssm2602_hw_params,
-		.shutdown = ssm2602_shutdown,
-		.digital_mute = ssm2602_mute,
-		.set_sysclk = ssm2602_set_dai_sysclk,
-		.set_fmt = ssm2602_set_dai_fmt,
-	}
+	.ops = &ssm2602_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ssm2602_dai);
 
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 8b20c360adf5..c3f4afb5d017 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -580,6 +580,15 @@ 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 = {
+	.prepare	= tlv320aic23_pcm_prepare,
+	.hw_params	= tlv320aic23_hw_params,
+	.shutdown	= tlv320aic23_shutdown,
+	.digital_mute	= tlv320aic23_mute,
+	.set_fmt	= tlv320aic23_set_dai_fmt,
+	.set_sysclk	= tlv320aic23_set_dai_sysclk,
+};
+
 struct snd_soc_dai tlv320aic23_dai = {
 	.name = "tlv320aic23",
 	.playback = {
@@ -594,14 +603,7 @@ struct snd_soc_dai tlv320aic23_dai = {
 		    .channels_max = 2,
 		    .rates = AIC23_RATES,
 		    .formats = AIC23_FORMATS,},
-	.ops = {
-		.prepare = tlv320aic23_pcm_prepare,
-		.hw_params = tlv320aic23_hw_params,
-		.shutdown = tlv320aic23_shutdown,
-		.digital_mute = tlv320aic23_mute,
-		.set_fmt = tlv320aic23_set_dai_fmt,
-		.set_sysclk = tlv320aic23_set_dai_sysclk,
-	}
+	.ops = &tlv320aic23_dai_ops,
 };
 EXPORT_SYMBOL_GPL(tlv320aic23_dai);
 
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 229e464cf713..a7f333fc579e 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -270,6 +270,13 @@ 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 = {
+	.hw_params	= aic26_hw_params,
+	.digital_mute	= aic26_mute,
+	.set_sysclk	= aic26_set_sysclk,
+	.set_fmt	= aic26_set_fmt,
+};
+
 struct snd_soc_dai aic26_dai = {
 	.name = "tlv320aic26",
 	.playback = {
@@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = {
 		.rates = AIC26_RATES,
 		.formats = AIC26_FORMATS,
 	},
-	.ops = {
-		.hw_params = aic26_hw_params,
-		.digital_mute = aic26_mute,
-		.set_sysclk = aic26_set_sysclk,
-		.set_fmt = aic26_set_fmt,
-	},
+	.ops = &aic26_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic26_dai);
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index d638e3f0728b..ab099f482487 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1088,6 +1088,13 @@ 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 = {
+	.hw_params	= aic3x_hw_params,
+	.digital_mute	= aic3x_mute,
+	.set_sysclk	= aic3x_set_dai_sysclk,
+	.set_fmt	= aic3x_set_dai_fmt,
+};
+
 struct snd_soc_dai aic3x_dai = {
 	.name = "tlv320aic3x",
 	.playback = {
@@ -1102,12 +1109,7 @@ struct snd_soc_dai aic3x_dai = {
 		.channels_max = 2,
 		.rates = AIC3X_RATES,
 		.formats = AIC3X_FORMATS,},
-	.ops = {
-		.hw_params = aic3x_hw_params,
-		.digital_mute = aic3x_mute,
-		.set_sysclk = aic3x_set_dai_sysclk,
-		.set_fmt = aic3x_set_dai_fmt,
-	}
+	.ops = &aic3x_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic3x_dai);
 
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 661599295ca7..ddefb8f80145 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -431,6 +431,15 @@ 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 = {
+	.startup	= uda134x_startup,
+	.shutdown	= uda134x_shutdown,
+	.hw_params	= uda134x_hw_params,
+	.digital_mute	= uda134x_mute,
+	.set_sysclk	= uda134x_set_dai_sysclk,
+	.set_fmt	= uda134x_set_dai_fmt,
+};
+
 struct snd_soc_dai uda134x_dai = {
 	.name = "UDA134X",
 	/* playback capabilities */
@@ -450,14 +459,7 @@ struct snd_soc_dai uda134x_dai = {
 		.formats = UDA134X_FORMATS,
 	},
 	/* pcm operations */
-	.ops = {
-		.startup = uda134x_startup,
-		.shutdown = uda134x_shutdown,
-		.hw_params = uda134x_hw_params,
-		.digital_mute = uda134x_mute,
-		.set_sysclk = uda134x_set_dai_sysclk,
-		.set_fmt = uda134x_set_dai_fmt,
-	}
+	.ops = &uda134x_dai_ops,
 };
 EXPORT_SYMBOL(uda134x_dai);
 
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 5242b8156b38..cafa7684c0e7 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -583,6 +583,29 @@ 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 = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.prepare	= uda1380_pcm_prepare,
+	.digital_mute	= uda1380_mute,
+	.set_fmt	= uda1380_set_dai_fmt_both,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_playback = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.prepare	= uda1380_pcm_prepare,
+	.digital_mute	= uda1380_mute,
+	.set_fmt	= uda1380_set_dai_fmt_playback,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_capture = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.prepare	= uda1380_pcm_prepare,
+	.set_fmt	= uda1380_set_dai_fmt_capture,
+};
+
 struct snd_soc_dai uda1380_dai[] = {
 {
 	.name = "UDA1380",
@@ -598,13 +621,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.channels_max = 2,
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt_both,
-	},
+	.ops = &uda1380_dai_ops,
 },
 { /* playback only - dual interface */
 	.name = "UDA1380",
@@ -615,13 +632,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt_playback,
-	},
+	.ops = &uda1380_dai_ops_playback,
 },
 { /* capture only - dual interface*/
 	.name = "UDA1380",
@@ -632,12 +643,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.set_fmt = uda1380_set_dai_fmt_capture,
-	},
+	.ops = &uda1380_dai_ops_capture,
 },
 };
 EXPORT_SYMBOL_GPL(uda1380_dai);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 359e5cc86f34..3b1d0993bed9 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1538,6 +1538,16 @@ static int wm8350_remove(struct platform_device *pdev)
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8350_dai_ops = {
+	 .hw_params	= wm8350_pcm_hw_params,
+	 .digital_mute	= wm8350_mute,
+	 .trigger	= wm8350_pcm_trigger,
+	 .set_fmt	= wm8350_set_dai_fmt,
+	 .set_sysclk	= wm8350_set_dai_sysclk,
+	 .set_pll	= wm8350_set_fll,
+	 .set_clkdiv	= wm8350_set_clkdiv,
+};
+
 struct snd_soc_dai wm8350_dai = {
 	.name = "WM8350",
 	.playback = {
@@ -1554,15 +1564,7 @@ struct snd_soc_dai wm8350_dai = {
 		 .rates = WM8350_RATES,
 		 .formats = WM8350_FORMATS,
 	 },
-	.ops = {
-		 .hw_params = wm8350_pcm_hw_params,
-		 .digital_mute = wm8350_mute,
-		 .trigger = wm8350_pcm_trigger,
-		 .set_fmt = wm8350_set_dai_fmt,
-		 .set_sysclk = wm8350_set_dai_sysclk,
-		 .set_pll = wm8350_set_fll,
-		 .set_clkdiv = wm8350_set_clkdiv,
-	 },
+	.ops = &wm8350_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8350_dai);
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index f01078cfbd72..cc975a62fa5c 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -554,6 +554,14 @@ 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 = {
+	.hw_params	= wm8510_pcm_hw_params,
+	.digital_mute	= wm8510_mute,
+	.set_fmt	= wm8510_set_dai_fmt,
+	.set_clkdiv	= wm8510_set_dai_clkdiv,
+	.set_pll	= wm8510_set_dai_pll,
+};
+
 struct snd_soc_dai wm8510_dai = {
 	.name = "WM8510 HiFi",
 	.playback = {
@@ -568,13 +576,7 @@ struct snd_soc_dai wm8510_dai = {
 		.channels_max = 2,
 		.rates = WM8510_RATES,
 		.formats = WM8510_FORMATS,},
-	.ops = {
-		.hw_params = wm8510_pcm_hw_params,
-		.digital_mute = wm8510_mute,
-		.set_fmt = wm8510_set_dai_fmt,
-		.set_clkdiv = wm8510_set_dai_clkdiv,
-		.set_pll = wm8510_set_dai_pll,
-	},
+	.ops = &wm8510_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8510_dai);
 
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d3c51ba5e6f9..ee0af23a1acc 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -771,6 +771,21 @@ 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 = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+	.digital_mute	= wm8580_digital_mute,
+};
+
+static struct snd_soc_dai_ops wm8580_dai_ops_capture = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+};
+
 struct snd_soc_dai wm8580_dai[] = {
 	{
 		.name = "WM8580 PAIFRX",
@@ -782,13 +797,7 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-			 .digital_mute = wm8580_digital_mute,
-		 },
+		.ops = &wm8580_dai_ops_playback,
 	},
 	{
 		.name = "WM8580 PAIFTX",
@@ -800,12 +809,7 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-		 },
+		.ops = &wm8580_dai_ops_capture,
 	},
 };
 EXPORT_SYMBOL_GPL(wm8580_dai);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index f8363b308895..e7ff2121ede9 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -244,6 +244,12 @@ 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 = {
+	.hw_params	= wm8728_hw_params,
+	.digital_mute	= wm8728_mute,
+	.set_fmt	= wm8728_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8728_dai = {
 	.name = "WM8728",
 	.playback = {
@@ -253,11 +259,7 @@ struct snd_soc_dai wm8728_dai = {
 		.rates = WM8728_RATES,
 		.formats = WM8728_FORMATS,
 	},
-	.ops = {
-		 .hw_params = wm8728_hw_params,
-		 .digital_mute = wm8728_mute,
-		 .set_fmt = wm8728_set_dai_fmt,
-	}
+	.ops = &wm8728_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8728_dai);
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 9e7ebcc2c491..e043e3f60008 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -433,6 +433,15 @@ 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 = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8731_dai = {
 	.name = "WM8731",
 	.playback = {
@@ -447,14 +456,7 @@ struct snd_soc_dai wm8731_dai = {
 		.channels_max = 2,
 		.rates = WM8731_RATES,
 		.formats = WM8731_FORMATS,},
-	.ops = {
-		.prepare = wm8731_pcm_prepare,
-		.hw_params = wm8731_hw_params,
-		.shutdown = wm8731_shutdown,
-		.digital_mute = wm8731_mute,
-		.set_sysclk = wm8731_set_dai_sysclk,
-		.set_fmt = wm8731_set_dai_fmt,
-	}
+	.ops = &wm8731_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8731_dai);
 
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 96afb86addc6..b64509b01a49 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -679,6 +679,13 @@ 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 = {
+	.hw_params	= wm8750_pcm_hw_params,
+	.digital_mute	= wm8750_mute,
+	.set_fmt	= wm8750_set_dai_fmt,
+	.set_sysclk	= wm8750_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8750_dai = {
 	.name = "WM8750",
 	.playback = {
@@ -693,12 +700,7 @@ struct snd_soc_dai wm8750_dai = {
 		.channels_max = 2,
 		.rates = WM8750_RATES,
 		.formats = WM8750_FORMATS,},
-	.ops = {
-		.hw_params = wm8750_pcm_hw_params,
-		.digital_mute = wm8750_mute,
-		.set_fmt = wm8750_set_dai_fmt,
-		.set_sysclk = wm8750_set_dai_sysclk,
-	},
+	.ops = &wm8750_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8750_dai);
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 7f353e935d71..cc6e57f9acf8 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1306,6 +1306,51 @@ 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_mode1 = {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1h_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1v_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode2_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
 static const struct snd_soc_dai wm8753_all_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "WM8753 HiFi",
@@ -1322,14 +1367,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1h_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode1,
 },
 /* DAI Voice mode 1 */
 {	.name = "WM8753 Voice",
@@ -1346,14 +1384,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1v_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode1,
 },
 /* DAI HiFi mode 2 - dummy */
 {	.name = "WM8753 HiFi",
@@ -1374,14 +1405,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode2_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode2,
 },
 /* DAI HiFi mode 3 */
 {	.name = "WM8753 HiFi",
@@ -1398,14 +1422,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode3,
 },
 /* DAI Voice mode 3 - dummy */
 {	.name = "WM8753 Voice",
@@ -1426,14 +1443,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode4,
 },
 /* DAI Voice mode 4 - dummy */
 {	.name = "WM8753 Voice",
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index da5ca64f89bb..46c5ea1ff921 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1088,6 +1088,14 @@ 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 = {
+	.hw_params	= wm8900_hw_params,
+	.set_clkdiv	= wm8900_set_dai_clkdiv,
+	.set_pll	= wm8900_set_dai_pll,
+	.set_fmt	= wm8900_set_dai_fmt,
+	.digital_mute	= wm8900_digital_mute,
+};
+
 struct snd_soc_dai wm8900_dai = {
 	.name = "WM8900 HiFi",
 	.playback = {
@@ -1104,13 +1112,7 @@ struct snd_soc_dai wm8900_dai = {
 		.rates = WM8900_RATES,
 		.formats = WM8900_PCM_FORMATS,
 	 },
-	.ops = {
-		.hw_params = wm8900_hw_params,
-		 .set_clkdiv = wm8900_set_dai_clkdiv,
-		 .set_pll = wm8900_set_dai_pll,
-		 .set_fmt = wm8900_set_dai_fmt,
-		 .digital_mute = wm8900_digital_mute,
-	 },
+	.ops = &wm8900_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8900_dai);
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index c6fa8a71b4dd..8cf571f1a803 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1497,6 +1497,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8903_dai_ops = {
+	.startup	= wm8903_startup,
+	.shutdown	= wm8903_shutdown,
+	.hw_params	= wm8903_hw_params,
+	.digital_mute	= wm8903_digital_mute,
+	.set_fmt	= wm8903_set_dai_fmt,
+	.set_sysclk	= wm8903_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8903_dai = {
 	.name = "WM8903",
 	.playback = {
@@ -1513,14 +1522,7 @@ struct snd_soc_dai wm8903_dai = {
 		 .rates = WM8903_CAPTURE_RATES,
 		 .formats = WM8903_FORMATS,
 	 },
-	.ops = {
-		 .startup = wm8903_startup,
-		 .shutdown = wm8903_shutdown,
-		 .hw_params = wm8903_hw_params,
-		 .digital_mute = wm8903_digital_mute,
-		 .set_fmt = wm8903_set_dai_fmt,
-		 .set_sysclk = wm8903_set_dai_sysclk
-	}
+	.ops = &wm8903_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8903_dai);
 
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 24d4c905a011..032dca22dbd3 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -604,6 +604,13 @@ 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 = {
+	.hw_params	= wm8971_pcm_hw_params,
+	.digital_mute	= wm8971_mute,
+	.set_fmt	= wm8971_set_dai_fmt,
+	.set_sysclk	= wm8971_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8971_dai = {
 	.name = "WM8971",
 	.playback = {
@@ -618,12 +625,7 @@ struct snd_soc_dai wm8971_dai = {
 		.channels_max = 2,
 		.rates = WM8971_RATES,
 		.formats = WM8971_FORMATS,},
-	.ops = {
-		.hw_params = wm8971_pcm_hw_params,
-		.digital_mute = wm8971_mute,
-		.set_fmt = wm8971_set_dai_fmt,
-		.set_sysclk = wm8971_set_dai_sysclk,
-	},
+	.ops = &wm8971_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8971_dai);
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 1a38421f7594..c518c3e5aa3f 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1332,6 +1332,15 @@ 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 = {
+	.hw_params	= wm8990_hw_params,
+	.digital_mute	= wm8990_mute,
+	.set_fmt	= wm8990_set_dai_fmt,
+	.set_clkdiv	= wm8990_set_dai_clkdiv,
+	.set_pll	= wm8990_set_dai_pll,
+	.set_sysclk	= wm8990_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8990_dai = {
 /* ADC/DAC on primary */
 	.name = "WM8990 ADC/DAC Primary",
@@ -1348,14 +1357,7 @@ struct snd_soc_dai wm8990_dai = {
 		.channels_max = 2,
 		.rates = WM8990_RATES,
 		.formats = WM8990_FORMATS,},
-	.ops = {
-		.hw_params = wm8990_hw_params,
-		.digital_mute = wm8990_mute,
-		.set_fmt = wm8990_set_dai_fmt,
-		.set_clkdiv = wm8990_set_dai_clkdiv,
-		.set_pll = wm8990_set_dai_pll,
-		.set_sysclk = wm8990_set_dai_sysclk,
-	},
+	.ops = &wm8990_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8990_dai);
 
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 2e9e06b2daaf..3265817c5c26 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -269,6 +269,10 @@ 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 = {
+	.prepare	= ac97_prepare,
+};
+
 struct snd_soc_dai wm9705_dai[] = {
 	{
 		.name = "AC97 HiFi",
@@ -287,9 +291,7 @@ struct snd_soc_dai wm9705_dai[] = {
 			.rates = WM9705_AC97_RATES,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
 		},
-		.ops = {
-			.prepare = ac97_prepare,
-		},
+		.ops = &wm9705_dai_ops,
 	},
 	{
 		.name = "AC97 Aux",
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index b3a8be77676e..765cf1e7369e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -517,6 +517,14 @@ 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 = {
+	.prepare	= ac97_prepare,
+};
+
+static struct snd_soc_dai_ops wm9712_dai_ops_aux = {
+	.prepare	= ac97_aux_prepare,
+};
+
 struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -533,8 +541,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 2,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &wm9712_dai_ops_hifi,
 },
 {
 	.name = "AC97 Aux",
@@ -544,8 +551,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 1,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,},
+	.ops = &wm9712_dai_ops_aux,
 }
 };
 EXPORT_SYMBOL_GPL(wm9712_dai);
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index a93aea5c1878..523bad077fa0 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1005,6 +1005,27 @@ 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 = {
+	.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 = {
+	.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 = {
+	.hw_params	= wm9713_pcm_hw_params,
+	.shutdown	= wm9713_voiceshutdown,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+	.set_fmt	= wm9713_set_dai_fmt,
+	.set_tristate	= wm9713_set_dai_tristate,
+};
+
 struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -1021,10 +1042,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_hifi_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_hifi,
 	},
 	{
 	.name = "AC97 Aux",
@@ -1034,10 +1052,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 1,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_aux,
 	},
 	{
 	.name = "WM9713 Voice",
@@ -1053,14 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
-	.ops = {
-		.hw_params = wm9713_pcm_hw_params,
-		.shutdown = wm9713_voiceshutdown,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,
-		.set_fmt = wm9713_set_dai_fmt,
-		.set_tristate = wm9713_set_dai_tristate,
-	},
+	.ops = &wm9713_dai_ops_voice,
 	},
 };
 EXPORT_SYMBOL_GPL(wm9713_dai);
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 0fee779e3c76..ffdb9439d3d8 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
+static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+	.startup	= davinci_i2s_startup,
+	.trigger	= davinci_i2s_trigger,
+	.hw_params	= davinci_i2s_hw_params,
+	.set_fmt	= davinci_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
@@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = {
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = davinci_i2s_startup,
-		.trigger = davinci_i2s_trigger,
-		.hw_params = davinci_i2s_hw_params,
-		.set_fmt = davinci_i2s_set_dai_fmt,
-	},
+	.ops = &davinci_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 6844009833db..0fddd437a7c9 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -562,6 +562,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * fsl_ssi_dai_template: template CPU DAI for the SSI
  */
+static struct snd_soc_dai_ops fsl_ssi_dai_ops = {
+	.startup	= fsl_ssi_startup,
+	.hw_params	= fsl_ssi_hw_params,
+	.shutdown	= fsl_ssi_shutdown,
+	.trigger	= fsl_ssi_trigger,
+	.set_sysclk	= fsl_ssi_set_sysclk,
+	.set_fmt	= fsl_ssi_set_fmt,
+};
+
 static struct snd_soc_dai fsl_ssi_dai_template = {
 	.playback = {
 		/* The SSI does not support monaural audio. */
@@ -576,14 +585,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = fsl_ssi_startup,
-		.hw_params = fsl_ssi_hw_params,
-		.shutdown = fsl_ssi_shutdown,
-		.trigger = fsl_ssi_trigger,
-		.set_sysclk = fsl_ssi_set_sysclk,
-		.set_fmt = fsl_ssi_set_fmt,
-	},
+	.ops = &fsl_ssi_dai_ops,
 };
 
 /**
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9eb1ce185bd0..3aa729df27b5 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -468,6 +468,16 @@ 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 = {
+	.startup	= psc_i2s_startup,
+	.hw_params	= psc_i2s_hw_params,
+	.hw_free	= psc_i2s_hw_free,
+	.shutdown	= psc_i2s_shutdown,
+	.trigger	= psc_i2s_trigger,
+	.set_sysclk	= psc_i2s_set_sysclk,
+	.set_fmt	= psc_i2s_set_fmt,
+};
+
 static struct snd_soc_dai psc_i2s_dai_template = {
 	.playback = {
 		.channels_min = 2,
@@ -481,15 +491,7 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.rates = PSC_I2S_RATES,
 		.formats = PSC_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = psc_i2s_startup,
-		.hw_params = psc_i2s_hw_params,
-		.hw_free = psc_i2s_hw_free,
-		.shutdown = psc_i2s_shutdown,
-		.trigger = psc_i2s_trigger,
-		.set_sysclk = psc_i2s_set_sysclk,
-		.set_fmt = psc_i2s_set_fmt,
-	},
+	.ops = &psc_i2s_dai_ops,
 };
 
 /* ---------------------------------------------------------------------
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 05dd5abcddf4..d6882be33452 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -461,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
+static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
+	.startup	= omap_mcbsp_dai_startup,
+	.shutdown	= omap_mcbsp_dai_shutdown,
+	.trigger	= omap_mcbsp_dai_trigger,
+	.hw_params	= omap_mcbsp_dai_hw_params,
+	.set_fmt	= omap_mcbsp_dai_set_dai_fmt,
+	.set_clkdiv	= omap_mcbsp_dai_set_clkdiv,
+	.set_sysclk	= omap_mcbsp_dai_set_dai_sysclk,
+};
+
 #define OMAP_MCBSP_DAI_BUILDER(link_id)				\
 {								\
 	.name = "omap-mcbsp-dai-"#link_id,			\
@@ -477,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
-	.ops = {						\
-		.startup = omap_mcbsp_dai_startup,		\
-		.shutdown = omap_mcbsp_dai_shutdown,		\
-		.trigger = omap_mcbsp_dai_trigger,		\
-		.hw_params = omap_mcbsp_dai_hw_params,		\
-		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
-		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
-		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
-	},							\
+	.ops = &omap_mcbsp_dai_ops,				\
 	.private_data = &mcbsp_data[(link_id)].bus_id,		\
 }
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 4a973ab710be..3e18064e86b2 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -784,6 +784,19 @@ static void pxa_ssp_remove(struct platform_device *pdev,
 			    SNDRV_PCM_FMTBIT_S24_LE |	\
 			    SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+	.startup	= pxa_ssp_startup,
+	.shutdown	= pxa_ssp_shutdown,
+	.trigger	= pxa_ssp_trigger,
+	.hw_params	= pxa_ssp_hw_params,
+	.set_sysclk	= pxa_ssp_set_dai_sysclk,
+	.set_clkdiv	= pxa_ssp_set_dai_clkdiv,
+	.set_pll	= pxa_ssp_set_dai_pll,
+	.set_fmt	= pxa_ssp_set_dai_fmt,
+	.set_tdm_slot	= pxa_ssp_set_dai_tdm_slot,
+	.set_tristate	= pxa_ssp_set_dai_tristate,
+};
+
 struct snd_soc_dai pxa_ssp_dai[] = {
 	{
 		.name = "pxa2xx-ssp1",
@@ -804,18 +817,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{	.name = "pxa2xx-ssp2",
 		.id = 1,
@@ -835,18 +837,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp3",
@@ -867,18 +858,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp4",
@@ -899,18 +879,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 };
 EXPORT_SYMBOL_GPL(pxa_ssp_dai);
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 812c2b4d3e07..11cd0f289c16 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -164,6 +164,10 @@ 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_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_params,
+};
+
 /*
  * There is only 1 physical AC97 interface for pxa2xx, but it
  * has extra fifo's that can be used for aux DACs and ADCs.
@@ -189,8 +193,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_params,},
+	.ops = &pxa_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-aux",
@@ -208,8 +211,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_aux_params,},
+	.ops = &pxa_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -221,8 +223,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_mic_params,},
+	.ops = &pxa_ac97_dai_ops,
 },
 };
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 83b59d7fe96e..e6c24408c5f9 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -304,6 +304,15 @@ static int pxa2xx_i2s_resume(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 = {
+	.startup	= pxa2xx_i2s_startup,
+	.shutdown	= pxa2xx_i2s_shutdown,
+	.trigger	= pxa2xx_i2s_trigger,
+	.hw_params	= pxa2xx_i2s_hw_params,
+	.set_fmt	= pxa2xx_i2s_set_dai_fmt,
+	.set_sysclk	= pxa2xx_i2s_set_dai_sysclk,
+};
+
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
@@ -319,14 +328,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.channels_max = 2,
 		.rates = PXA2XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = pxa2xx_i2s_startup,
-		.shutdown = pxa2xx_i2s_shutdown,
-		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,
-		.set_fmt = pxa2xx_i2s_set_dai_fmt,
-		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
-	},
+	.ops = &pxa_i2s_dai_ops,
 };
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index f3fc0aba0aaf..382d7eee53ef 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -708,6 +708,14 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *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 s3c2412_i2s_dai_ops = {
+	.trigger	= s3c2412_i2s_trigger,
+	.hw_params	= s3c2412_i2s_hw_params,
+	.set_fmt	= s3c2412_i2s_set_fmt,
+	.set_clkdiv	= s3c2412_i2s_set_clkdiv,
+	.set_sysclk	= s3c2412_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c2412_i2s_dai = {
 	.name	= "s3c2412-i2s",
 	.id	= 0,
@@ -726,13 +734,7 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 		.rates		= S3C2412_I2S_RATES,
 		.formats	= SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.trigger	= s3c2412_i2s_trigger,
-		.hw_params	= s3c2412_i2s_hw_params,
-		.set_fmt	= s3c2412_i2s_set_fmt,
-		.set_clkdiv	= s3c2412_i2s_set_clkdiv,
-		.set_sysclk	= s3c2412_i2s_set_sysclk,
-	},
+	.ops = &s3c2412_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 5822d2dd49ba..83ea623234e7 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -355,6 +355,11 @@ static int s3c2443_ac97_mic_trigger(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 s3c2443_ac97_dai_ops = {
+	.hw_params	= s3c2443_ac97_hw_params,
+	.trigger	= s3c2443_ac97_trigger,
+};
+
 struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
@@ -374,9 +379,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_params,
-		.trigger = s3c2443_ac97_trigger},
+	.ops = &s3c2443_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -388,9 +391,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_mic_params,
-		.trigger = s3c2443_ac97_mic_trigger,},
+	.ops = &s3c2443_ac97_dai_ops,
 },
 };
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 1c2b05497107..4473fb584c4c 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -456,6 +456,14 @@ 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 = {
+	.trigger	= s3c24xx_i2s_trigger,
+	.hw_params	= s3c24xx_i2s_hw_params,
+	.set_fmt	= s3c24xx_i2s_set_fmt,
+	.set_clkdiv	= s3c24xx_i2s_set_clkdiv,
+	.set_sysclk	= s3c24xx_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
@@ -472,13 +480,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = S3C24XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.trigger = s3c24xx_i2s_trigger,
-		.hw_params = s3c24xx_i2s_hw_params,
-		.set_fmt = s3c24xx_i2s_set_fmt,
-		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
-		.set_sysclk = s3c24xx_i2s_set_sysclk,
-	},
+	.ops = &s3c24xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index d1e5390fddeb..56fa0872abbb 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -336,6 +336,16 @@ 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 = {
+	.startup	= ssi_startup,
+	.shutdown	= ssi_shutdown,
+	.trigger	= ssi_trigger,
+	.hw_params	= ssi_hw_params,
+	.set_sysclk	= ssi_set_sysclk,
+	.set_clkdiv	= ssi_set_clkdiv,
+	.set_fmt	= ssi_set_fmt,
+};
+
 struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
@@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
@@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #endif
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d4b90d82a098..16518329f6b2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	mutex_lock(&pcm_mutex);
 
 	/* startup the audio subsystem */
-	if (cpu_dai->ops.startup) {
-		ret = cpu_dai->ops.startup(substream, cpu_dai);
+	if (cpu_dai->ops->startup) {
+		ret = cpu_dai->ops->startup(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open interface %s\n",
 				cpu_dai->name);
@@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.startup) {
-		ret = codec_dai->ops.startup(substream, codec_dai);
+	if (codec_dai->ops->startup) {
+		ret = codec_dai->ops->startup(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open codec %s\n",
 				codec_dai->name);
@@ -247,8 +247,8 @@ codec_dai_err:
 		platform->pcm_ops->close(substream);
 
 platform_err:
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		snd_soc_dai_digital_mute(codec_dai, 1);
 
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 
-	if (codec_dai->ops.shutdown)
-		codec_dai->ops.shutdown(substream, codec_dai);
+	if (codec_dai->ops->shutdown)
+		codec_dai->ops->shutdown(substream, codec_dai);
 
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
@@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.prepare) {
-		ret = codec_dai->ops.prepare(substream, codec_dai);
+	if (codec_dai->ops->prepare) {
+		ret = codec_dai->ops->prepare(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: codec DAI prepare error\n");
 			goto out;
 		}
 	}
 
-	if (cpu_dai->ops.prepare) {
-		ret = cpu_dai->ops.prepare(substream, cpu_dai);
+	if (cpu_dai->ops->prepare) {
+		ret = cpu_dai->ops->prepare(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
 			goto out;
@@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (codec_dai->ops.hw_params) {
-		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
+	if (codec_dai->ops->hw_params) {
+		ret = codec_dai->ops->hw_params(substream, params, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
 				codec_dai->name);
@@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (cpu_dai->ops.hw_params) {
-		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
+	if (cpu_dai->ops->hw_params) {
+		ret = cpu_dai->ops->hw_params(substream, params, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
@@ -526,12 +526,12 @@ out:
 	return ret;
 
 platform_err:
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 interface_err:
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
 codec_err:
 	if (machine->ops && machine->ops->hw_free)
@@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		platform->pcm_ops->hw_free(substream);
 
 	/* now free hw params for the DAI's  */
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
 
-	if (codec_dai->ops.trigger) {
-		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
+	if (codec_dai->ops->trigger) {
+		ret = codec_dai->ops->trigger(substream, cmd, codec_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 			return ret;
 	}
 
-	if (cpu_dai->ops.trigger) {
-		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
+	if (cpu_dai->ops->trigger) {
+		ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	/* mute any active DAC's */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
-		if (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 1);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
@@ -741,8 +741,8 @@ static void soc_resume_deferred(struct work_struct *work)
 	/* unmute any active DACs */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
-		if (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 0);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 0);
 	}
 
 	for (i = 0; i < card->num_links; i++) {
@@ -2051,8 +2051,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
 }
@@ -2071,8 +2071,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->ops.set_clkdiv)
-		return dai->ops.set_clkdiv(dai, div_id, div);
+	if (dai->ops->set_clkdiv)
+		return dai->ops->set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
 }
@@ -2090,8 +2090,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->ops.set_pll)
-		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
+	if (dai->ops->set_pll)
+		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -2106,8 +2106,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->ops.set_fmt)
-		return dai->ops.set_fmt(dai, fmt);
+	if (dai->ops->set_fmt)
+		return dai->ops->set_fmt(dai, fmt);
 	else
 		return -EINVAL;
 }
@@ -2125,8 +2125,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tdm_slot(dai, mask, slots);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
 }
@@ -2141,8 +2141,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tristate(dai, tristate);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tristate(dai, tristate);
 	else
 		return -EINVAL;
 }
@@ -2157,8 +2157,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->ops.digital_mute)
-		return dai->ops.digital_mute(dai, mute);
+	if (dai->ops->digital_mute)
+		return dai->ops->digital_mute(dai, mute);
 	else
 		return -EINVAL;
 }
@@ -2211,6 +2211,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 	return 0;
 }
 
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
 /**
  * snd_soc_register_dai - Register a DAI with the ASoC core
  *
@@ -2225,6 +2228,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai)
 	if (!dai->dev)
 		printk(KERN_WARNING "No device for DAI %s\n", dai->name);
 
+	if (!dai->ops)
+		dai->ops = &null_dai_ops;
+
 	INIT_LIST_HEAD(&dai->list);
 
 	mutex_lock(&client_mutex);