diff options
author | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2009-05-24 12:32:24 (GMT) |
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committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2009-05-24 12:32:24 (GMT) |
commit | 05e1efa2deb42b1bd548208e5c43f471e2cf0da1 (patch) | |
tree | 9242ae7aa647c7ae5347994bf134d56ecf53b1a8 /sound | |
parent | 3c166c7f1828f226c7f478758bf6c8ce8be1623f (diff) | |
download | linux-fsl-qoriq-05e1efa2deb42b1bd548208e5c43f471e2cf0da1.tar.xz |
ASoC: Fix minor issues in STAC9766 driver
Fairly minor issues:
- Don't register the DAIs, it's not required for AC97 devices.
- Make unexported functions static.
- Wrap some excessively long lines.
- Undo tab/space breakage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/codecs/stac9766.c | 65 |
1 files changed, 29 insertions, 36 deletions
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 7740cd5..8ad4b7b 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -52,12 +52,14 @@ static const u16 stac9766_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ }; -static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; -static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_record_all_mux[] = {"All analog", + "Analog plus DAC"}; static const char *stac9766_boost1[] = {"0dB", "10dB"}; static const char *stac9766_boost2[] = {"0dB", "20dB"}; static const char *stac9766_stereo_mic[] = {"Off", "On"}; @@ -73,7 +75,8 @@ static const struct soc_enum stac9766_SPDIF_enum = static const struct soc_enum stac9766_popbypass_enum = SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); static const struct soc_enum stac9766_record_all_enum = - SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, + stac9766_record_all_mux); static const struct soc_enum stac9766_boost1_enum = SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ static const struct soc_enum stac9766_boost2_enum = @@ -89,9 +92,11 @@ static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), - SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, + master_tlv), SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), - SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, + master_tlv), SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), @@ -133,8 +138,8 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), }; -int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) +static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) { u16 *cache = codec->reg_cache; @@ -152,7 +157,8 @@ int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, + unsigned int reg) { u16 val = 0, *cache = codec->reg_cache; @@ -176,7 +182,7 @@ unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) } static int ac97_analog_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; @@ -197,7 +203,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, } static int ac97_digital_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; @@ -216,7 +222,7 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, } static int ac97_digital_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) + int cmd, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; unsigned short vra; @@ -232,7 +238,7 @@ static int ac97_digital_trigger(struct snd_pcm_substream *substream, } static int stac9766_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: /* full On */ @@ -249,7 +255,7 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, return 0; } -int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); @@ -266,7 +272,7 @@ int stac9766_reset(struct snd_soc_codec *codec, int try_warm) } static int stac9766_codec_suspend(struct platform_device *pdev, - pm_message_t state) + pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -303,13 +309,11 @@ reset: return 0; } -static struct snd_soc_dai_ops stac9766_dai_ops_analog = -{ +static struct snd_soc_dai_ops stac9766_dai_ops_analog = { .prepare = ac97_analog_prepare, }; -static struct snd_soc_dai_ops stac9766_dai_ops_digital = -{ +static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, .trigger = ac97_digital_trigger, }; @@ -354,7 +358,8 @@ struct snd_soc_dai stac9766_dai[] = { }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, -}}; +} +}; EXPORT_SYMBOL_GPL(stac9766_dai); static int stac9766_codec_probe(struct platform_device *pdev) @@ -371,7 +376,8 @@ static int stac9766_codec_probe(struct platform_device *pdev) codec = socdev->card->codec; mutex_init(&codec->mutex); - codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), + GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; @@ -409,8 +415,8 @@ static int stac9766_codec_probe(struct platform_device *pdev) stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( - stac9766_snd_ac97_controls)); + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, + ARRAY_SIZE(stac9766_snd_ac97_controls)); ret = snd_soc_init_card(socdev); if (ret < 0) @@ -444,8 +450,7 @@ static int stac9766_codec_remove(struct platform_device *pdev) return 0; } -struct snd_soc_codec_device soc_codec_dev_stac9766 = -{ +struct snd_soc_codec_device soc_codec_dev_stac9766 = { .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, @@ -453,18 +458,6 @@ struct snd_soc_codec_device soc_codec_dev_stac9766 = }; EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); -static int __init stac9766_modinit(void) -{ - return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); -} -module_init(stac9766_modinit); - -static void __exit stac9766_exit(void) -{ - snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); -} -module_exit(stac9766_exit); - MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); MODULE_LICENSE("GPL"); |