Merge branch 'patches_for_2.6.38rc' of git://git.pwsan.com/linux-2.6 into devel-fixes

19 files changed, 73 insertions, 46 deletions

diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e06b0d..5e4d499d8434 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
         .cpu_dai_name = "atmel-ssc-dai.0",
         .codec_dai_name = "tlv320aic23-hifi",
         .platform_name = "atmel_pcm-audio",
-        .codec_name = "tlv320aic23-codec.0-0x1a",
+        .codec_name = "tlv320aic23-codec.0-001a",
         .init = afeb9260_tlv320aic23_init,
         .ops = &afeb9260_ops,
 };
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24c1856..ad28663f5bbd 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
         .cpu_dai_name = "bf5xx-i2s",
         .codec_dai_name = "ssm2602-hifi",
         .platform_name = "bf5xx-pcm-audio",
-        .codec_name = "ssm2602-codec.0-0x1b",
+        .codec_name = "ssm2602-codec.0-001b",
         .ops = &bf5xx_ssm2602_ops,
 };
 
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd067f79..347a567b01e1 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
 
 static int cq93vc_probe(struct snd_soc_codec *codec)
 {
-        struct davinci_vc *davinci_vc = codec->dev->platform_data;
+        struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
 
         davinci_vc->cq93vc.codec = codec;
         codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e860d229..bb4bf65b9e7e 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
         return 0;
 }
 
+static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+
 static struct snd_soc_codec_driver cx20442_codec_dev = {
         .probe =        cx20442_codec_probe,
         .remove =       cx20442_codec_remove,
+        .reg_cache_default = &cx20442_reg,
         .reg_cache_size = 1,
         .reg_word_size = sizeof(u8),
         .read = cx20442_read_reg_cache,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 247a6a99feb8..37b8aa8a680f 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
                       WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
                       WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
                       SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
                       WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = {
         { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
         { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
+        { "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+        { "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+        { "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+        { "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
         { "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
 
         /* AIF3 output */
@@ -1578,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = {
         { "Right Headphone Mux", "DAC", "DAC1R" },
 };
 
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+        { "AIF1DACDAT", NULL, "AIF2DACDAT" },
+        { "AIF2DACDAT", NULL, "AIF1DACDAT" },
+        { "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+        { "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+};
+
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
         { "AIF2DACL", NULL, "AIF2DAC Mux" },
         { "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -2386,7 +2399,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
         else
                 val = 0;
 
-        return snd_soc_update_bits(codec, reg, mask, reg);
+        return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 #define WM8994_RATES SNDRV_PCM_RATE_8000_96000
@@ -3129,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
         case WM8994:
                 snd_soc_dapm_add_routes(dapm, wm8994_intercon,
                                         ARRAY_SIZE(wm8994_intercon));
+
+                if (wm8994->revision < 4)
+                        snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+                                                ARRAY_SIZE(wm8994_revd_intercon));
+                        
                 break;
         case WM8958:
                 snd_soc_dapm_add_routes(dapm, wm8958_intercon,
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 6045cbde492b..608c84c5aa8e 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
         else
                 val = 0;
 
-        return snd_soc_update_bits(codec, reg, mask, reg);
+        return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 /* The size in bits of the FLL divide multiplied by 10
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index c466982eed23..613df5db0b32 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
         struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+        s8 offset;
         u16 reg, reg_l, reg_r, dcs_cfg;
 
         /* If we're using a digital only path and have a previously
@@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
                         hubs->dcs_codes);
 
                 /* HPOUT1L */
-                if (reg_l + hubs->dcs_codes > 0 &&
-                    reg_l + hubs->dcs_codes < 0xff)
-                        reg_l += hubs->dcs_codes;
-                dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+                offset = reg_l;
+                offset += hubs->dcs_codes;
+                dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
                 /* HPOUT1R */
-                if (reg_r + hubs->dcs_codes > 0 &&
-                    reg_r + hubs->dcs_codes < 0xff)
-                        reg_r += hubs->dcs_codes;
-                dcs_cfg |= reg_r;
+                offset = reg_r;
+                offset += hubs->dcs_codes;
+                dcs_cfg |= (u8)offset;
 
                 dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
 
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6bacc681..fe7984221eb9 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,12 +218,24 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
                 .ops = &evm_spdif_ops,
         },
 };
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+        .name = "TLV320AIC3X",
+        .stream_name = "AIC3X",
+        .cpu_dai_name = "davinci-mcasp.1",
+        .codec_dai_name = "tlv320aic3x-hifi",
+        .codec_name = "tlv320aic3x-codec.1-0018",
+        .platform_name = "davinci-pcm-audio",
+        .init = evm_aic3x_init,
+        .ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
         .name = "TLV320AIC3X",
         .stream_name = "AIC3X",
         .cpu_dai_name= "davinci-mcasp.0",
         .codec_dai_name = "tlv320aic3x-hifi",
-        .codec_name = "tlv320aic3x-codec.0-001a",
+        .codec_name = "tlv320aic3x-codec.1-0018",
         .platform_name = "davinci-pcm-audio",
         .init = evm_aic3x_init,
         .ops = &evm_ops,
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
 
 static struct snd_soc_card da830_snd_soc_card = {
         .name = "DA830/OMAP-L137 EVM",
-        .dai_link = &da8xx_evm_dai,
+        .dai_link = &da830_evm_dai,
         .num_links = 1,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
         .name = "DA850/OMAP-L138 EVM",
-        .dai_link = &da8xx_evm_dai,
+        .dai_link = &da850_evm_dai,
         .num_links = 1,
 };
 
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdcee21f..3167be689621 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
         /* Set up digital mute if not provided by the codec */
         if (!codec_dai->driver->ops) {
                 codec_dai->driver->ops = &ams_delta_dai_ops;
-        } else if (!codec_dai->driver->ops->digital_mute) {
-                codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
         } else {
                 ams_delta_ops.startup = ams_delta_startup;
                 ams_delta_ops.shutdown = ams_delta_shutdown;
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0d5fc7..784cff5f67e8 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link corgi_dai = {
         .name = "WM8731",
         .stream_name = "WM8731",
-        .cpu_dai_name = "pxa-is2-dai",
+        .cpu_dai_name = "pxa2xx-i2s",
         .codec_dai_name = "wm8731-hifi",
         .platform_name = "pxa-pcm-audio",
-        .codec_name = "wm8731-codec-0.001a",
+        .codec_name = "wm8731-codec-0.001b",
         .init = corgi_wm8731_init,
         .ops = &corgi_ops,
 };
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee115e27..a7d4999f9b24 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
         .cpu_dai_name = "pxa2xx-i2s",
         .codec_dai_name = "wm8731-hifi",
         .platform_name = "pxa-pcm-audio",
-        .codec_name = "wm8731-codec.0-001a",
+        .codec_name = "wm8731-codec.0-001b",
         .init = poodle_wm8731_init,
         .ops = &poodle_ops,
 };
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69b957a..8e1571350630 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link spitz_dai = {
         .name = "wm8750",
         .stream_name = "WM8750",
-        .cpu_dai_name = "pxa-is2",
+        .cpu_dai_name = "pxa2xx-i2s",
         .codec_dai_name = "wm8750-hifi",
         .platform_name = "pxa-pcm-audio",
-        .codec_name = "wm8750-codec.0-001a",
+        .codec_name = "wm8750-codec.0-001b",
         .init = spitz_wm8750_init,
         .ops = &spitz_ops,
 };
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 3eec610c10f9..0d0ae2b9eef6 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -397,11 +397,11 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
 { /* Hifi Playback - for similatious use with voice below */
         .name = "WM8753",
         .stream_name = "WM8753 HiFi",
-        .cpu_dai_name = "s3c24xx-i2s",
+        .cpu_dai_name = "s3c24xx-iis",
         .codec_dai_name = "wm8753-hifi",
         .init = neo1973_gta02_wm8753_init,
         .platform_name = "samsung-audio",
-        .codec_name = "wm8753-codec.0-0x1a",
+        .codec_name = "wm8753-codec.0-001a",
         .ops = &neo1973_gta02_hifi_ops,
 },
 { /* Voice via BT */
@@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
         .cpu_dai_name = "bluetooth-dai",
         .codec_dai_name = "wm8753-voice",
         .ops = &neo1973_gta02_voice_ops,
-        .codec_name = "wm8753-codec.0-0x1a",
+        .codec_name = "wm8753-codec.0-001a",
         .platform_name = "samsung-audio",
 },
 };
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a24514beb5..d20815d5ab2e 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -559,9 +559,9 @@ static struct snd_soc_dai_link neo1973_dai[] = {
         .name = "WM8753",
         .stream_name = "WM8753 HiFi",
         .platform_name = "samsung-audio",
-        .cpu_dai_name = "s3c24xx-i2s",
+        .cpu_dai_name = "s3c24xx-iis",
         .codec_dai_name = "wm8753-hifi",
-        .codec_name = "wm8753-codec.0-0x1a",
+        .codec_name = "wm8753-codec.0-001a",
         .init = neo1973_wm8753_init,
         .ops = &neo1973_hifi_ops,
 },
@@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
         .platform_name = "samsung-audio",
         .cpu_dai_name = "bluetooth-dai",
         .codec_dai_name = "wm8753-voice",
-        .codec_name = "wm8753-codec.0-0x1a",
+        .codec_name = "wm8753-codec.0-001a",
         .ops = &neo1973_voice_ops,
 },
 };
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index bb4292e3596c..08fcaaa66907 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -94,8 +94,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic33 = {
         .name           = "tlv320aic33",
         .stream_name    = "TLV320AIC33",
-        .codec_name     = "tlv320aic3x-codec.0-0x1a",
-        .cpu_dai_name   = "s3c24xx-i2s",
+        .codec_name     = "tlv320aic3x-codec.0-001a",
+        .cpu_dai_name   = "s3c24xx-iis",
         .codec_dai_name = "tlv320aic3x-hifi",
         .platform_name  = "samsung-audio",
         .init           = simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index fbba4e367729..116e3e670167 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -85,8 +85,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic23 = {
         .name           = "tlv320aic23",
         .stream_name    = "TLV320AIC23",
-        .codec_name     = "tlv320aic3x-codec.0-0x1a",
-        .cpu_dai_name   = "s3c24xx-i2s",
+        .codec_name     = "tlv320aic3x-codec.0-001a",
+        .cpu_dai_name   = "s3c24xx-iis",
         .codec_dai_name = "tlv320aic3x-hifi",
         .platform_name  = "samsung-audio",
         .init           = simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index cdc8ecbcb8ef..2c09e93dd566 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
         .stream_name = "UDA134X",
         .codec_name = "uda134x-hifi",
         .codec_dai_name = "uda134x-hifi",
-        .cpu_dai_name = "s3c24xx-i2s",
+        .cpu_dai_name = "s3c24xx-iis",
         .ops = &s3c24xx_uda134x_ops,
         .platform_name  = "samsung-audio",
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bac7291b6ff6..c3f6f1e72790 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
                 rtd = &card->rtd_aux[num];
                 name = aux_dev->name;
         }
+        rtd->card = card;
 
         /* machine controls, routes and widgets are not prefixed */
         temp = codec->name_prefix;
@@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 
         /* register the rtd device */
         rtd->codec = codec;
-        rtd->card = card;
         rtd->dev.parent = card->dev;
         rtd->dev.release = rtd_release;
         rtd->dev.init_name = name;
@@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
         goto out;
 
 found:
-        if (!try_module_get(codec->dev->driver->owner))
-                return -ENODEV;
-
         ret = soc_probe_codec(card, codec);
         if (ret < 0)
                 return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730ab5638..8194f150bab7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
         int max = mc->max;
         unsigned int mask = (1 << fls(max)) - 1;
         unsigned int invert = mc->invert;
-        unsigned int val, val_mask;
+        unsigned int val;
         int connect, change;
         struct snd_soc_dapm_update update;
 
@@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 
         if (invert)
                 val = max - val;
-        val_mask = mask << shift;
+        mask = mask << shift;
         val = val << shift;
 
         mutex_lock(&widget->codec->mutex);
         widget->value = val;
 
-        change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+        change = snd_soc_test_bits(widget->codec, reg, mask, val);
         if (change) {
                 if (val)
                         /* new connection */