From d77c290af76637e87dc07df28536231ee5042c98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:07:32 +0100 Subject: ASoC: tlv320dac33: Use SOC_ENUM_SINGLE_*_DECL() Just replace with the helper macros. No functional change at all. Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 32 +++++++++++++------------------- 1 file changed, 13 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs/tlv320dac33.c') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4f358393d6d..9ce849623a0 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -478,9 +478,7 @@ static const char *dac33_fifo_mode_texts[] = { "Bypass", "Mode 1", "Mode 7" }; -static const struct soc_enum dac33_fifo_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts), - dac33_fifo_mode_texts); +static SOC_ENUM_SINGLE_EXT_DECL(dac33_fifo_mode_enum, dac33_fifo_mode_texts); /* L/R Line Output Gain */ static const char *lr_lineout_gain_texts[] = { @@ -488,15 +486,13 @@ static const char *lr_lineout_gain_texts[] = { "Line 0dB DAC 12dB", "Line 6dB DAC 18dB", }; -static const struct soc_enum l_lineout_gain_enum = - SOC_ENUM_SINGLE(DAC33_LDAC_PWR_CTRL, 0, - ARRAY_SIZE(lr_lineout_gain_texts), - lr_lineout_gain_texts); +static SOC_ENUM_SINGLE_DECL(l_lineout_gain_enum, + DAC33_LDAC_PWR_CTRL, 0, + lr_lineout_gain_texts); -static const struct soc_enum r_lineout_gain_enum = - SOC_ENUM_SINGLE(DAC33_RDAC_PWR_CTRL, 0, - ARRAY_SIZE(lr_lineout_gain_texts), - lr_lineout_gain_texts); +static SOC_ENUM_SINGLE_DECL(r_lineout_gain_enum, + DAC33_RDAC_PWR_CTRL, 0, + lr_lineout_gain_texts); /* * DACL/R digital volume control: @@ -534,18 +530,16 @@ static const struct snd_kcontrol_new dac33_dapm_abypassr_control = /* LOP L/R invert selection */ static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"}; -static const struct soc_enum dac33_left_lom_enum = - SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3, - ARRAY_SIZE(dac33_lr_lom_texts), - dac33_lr_lom_texts); +static SOC_ENUM_SINGLE_DECL(dac33_left_lom_enum, + DAC33_OUT_AMP_CTRL, 3, + dac33_lr_lom_texts); static const struct snd_kcontrol_new dac33_dapm_left_lom_control = SOC_DAPM_ENUM("Route", dac33_left_lom_enum); -static const struct soc_enum dac33_right_lom_enum = - SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2, - ARRAY_SIZE(dac33_lr_lom_texts), - dac33_lr_lom_texts); +static SOC_ENUM_SINGLE_DECL(dac33_right_lom_enum, + DAC33_OUT_AMP_CTRL, 2, + dac33_lr_lom_texts); static const struct snd_kcontrol_new dac33_dapm_right_lom_control = SOC_DAPM_ENUM("Route", dac33_right_lom_enum); -- cgit v1.2.3-70-g09d2 From 5c898e74d135a23ce12e0263c1a3c78eeae1b52b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Mar 2014 13:17:45 +0100 Subject: ASoC: Add helper function to check whether a CODEC is active Instead of directly checking the 'active' field of the CODEC struct add a new helper function that will return either true or false depending on whether the CODEC is active. This will make the migration to the component level easier. The patch also updates all CODEC drivers that check the active attribute to use the new helper function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8753.c | 4 ++-- 8 files changed, 14 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs/tlv320dac33.c') diff --git a/include/sound/soc.h b/include/sound/soc.h index 53d15e0e6e8..5c2b4f4b5cf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1172,6 +1172,11 @@ static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) return 1; } +static inline bool snd_soc_codec_is_active(struct snd_soc_codec *codec) +{ + return codec->active != 0; +} + int snd_soc_util_init(void); void snd_soc_util_exit(void); diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f78b27a7c46..d50cf5b29a2 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -722,7 +722,7 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - if (!codec->active || !adav80x->rate) + if (!snd_soc_codec_is_active(codec) || !adav80x->rate) return 0; return snd_pcm_hw_constraint_minmax(substream->runtime, @@ -735,7 +735,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - if (!codec->active) + if (!snd_soc_codec_is_active(codec)) adav80x->rate = 0; } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 5d430cc56f5..458a6aed203 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -400,7 +400,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ - if (!codec->active) { + if (!snd_soc_codec_is_active(codec)) { udelay(50); snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4f358393d6d..35b2d244e42 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -461,7 +461,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, if (dac33->fifo_mode == ucontrol->value.integer.value[0]) return 0; /* Do not allow changes while stream is running*/ - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 726df6d43c2..8e3940dcff2 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -108,7 +108,7 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, /* the interpolator & decimator regs must only be written when the * codec DAI is active. */ - if (!codec->active && (reg >= UDA1380_MVOL)) + if (!snd_soc_codec_is_active(codec) && (reg >= UDA1380_MVOL)) return 0; pr_debug("uda1380: hw write %x val %x\n", reg, value); if (codec->hw_write(codec->control_data, data, 3) == 3) { diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index b7ab2ef567c..47e96ff3006 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -197,7 +197,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, return 0; /* Do not allow changes while stream is running */ - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index d99f948c513..6efcc40a7cb 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -201,7 +201,7 @@ static void wm8711_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; /* deactivate */ - if (!codec->active) { + if (!snd_soc_codec_is_active(codec)) { udelay(50); snd_soc_write(codec, WM8711_ACTIVE, 0x0); } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da93a26..5cf4bebc5d8 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -251,7 +251,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, if (wm8753->dai_func == ucontrol->value.integer.value[0]) return 0; - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EBUSY; ioctl = snd_soc_read(codec, WM8753_IOCTL); @@ -1314,7 +1314,7 @@ static int wm8753_mute(struct snd_soc_dai *dai, int mute) /* the digital mute covers the HiFi and Voice DAC's on the WM8753. * make sure we check if they are not both active when we mute */ if (mute && wm8753->dai_func == 1) { - if (!codec->active) + if (!snd_soc_codec_is_active(codec)) snd_soc_write(codec, WM8753_DAC, mute_reg | 0x8); } else { if (mute) -- cgit v1.2.3-70-g09d2