[PATCH v2 11/21] ASoC: codecs: use snd_soc_xxx_active()
Kuninori Morimoto
kuninori.morimoto.gx at renesas.com
Thu May 14 10:41:54 CEST 2020
From: Kuninori Morimoto <kuninori.morimoto.gx at renesas.com>
We have snd_soc_dai/dai_stream/component_active() macro
This patch uses it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx at renesas.com>
---
v1 -> v2
- adjust to latest patch-set
sound/soc/codecs/adav80x.c | 4 ++--
sound/soc/codecs/arizona.c | 2 +-
sound/soc/codecs/cs4271.c | 4 ++--
sound/soc/codecs/madera.c | 2 +-
sound/soc/codecs/max98090.c | 6 +++---
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 ++--
11 files changed, 16 insertions(+), 16 deletions(-)
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 7cea398ec392..c4b9722c3d8f 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -725,7 +725,7 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
struct adav80x *adav80x = snd_soc_component_get_drvdata(component);
- if (!snd_soc_component_is_active(component) || !adav80x->rate)
+ if (!snd_soc_component_active(component) || !adav80x->rate)
return 0;
return snd_pcm_hw_constraint_single(substream->runtime,
@@ -738,7 +738,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
struct adav80x *adav80x = snd_soc_component_get_drvdata(component);
- if (!snd_soc_component_is_active(component))
+ if (!snd_soc_component_active(component))
adav80x->rate = 0;
}
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 70341b30f567..9716c9624a89 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1926,7 +1926,7 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
if (clk_id == dai_priv->clk)
return 0;
- if (dai->active) {
+ if (snd_soc_dai_active(dai)) {
dev_err(component->dev, "Can't change clock on active DAI %d\n",
dai->id);
return -EBUSY;
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 62f412d6f9f2..d43762ae8f3d 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -356,9 +356,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
*/
if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
- !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) ||
+ !snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_CAPTURE)) ||
(substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
- !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) {
+ !snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_PLAYBACK))) {
ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
CS4271_MODE2_PDN,
CS4271_MODE2_PDN);
diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c
index a448d2a2918a..ec380b0b2d4e 100644
--- a/sound/soc/codecs/madera.c
+++ b/sound/soc/codecs/madera.c
@@ -3279,7 +3279,7 @@ static int madera_dai_set_sysclk(struct snd_soc_dai *dai,
if (is_sync == madera_is_syncclk(dai_priv->clk))
return 0;
- if (dai->active) {
+ if (snd_soc_dai_active(dai)) {
dev_err(component->dev, "Can't change clock on active DAI %d\n",
dai->id);
return -EBUSY;
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 032adc14562d..e2cc1ad8cb0a 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2039,7 +2039,7 @@ static int max98090_dai_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!max98090->master && dai->active == 1)
+ if (!max98090->master && snd_soc_dai_active(dai) == 1)
queue_delayed_work(system_power_efficient_wq,
&max98090->pll_det_enable_work,
msecs_to_jiffies(10));
@@ -2047,7 +2047,7 @@ static int max98090_dai_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- if (!max98090->master && dai->active == 1)
+ if (!max98090->master && snd_soc_dai_active(dai) == 1)
schedule_work(&max98090->pll_det_disable_work);
break;
default:
@@ -2109,7 +2109,7 @@ static void max98090_pll_work(struct max98090_priv *max98090)
unsigned int pll;
int i;
- if (!snd_soc_component_is_active(component))
+ if (!snd_soc_component_active(component))
return;
dev_info_ratelimited(component->dev, "PLL unlocked\n");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index f8e2f4b74db3..9868fb22323c 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -394,7 +394,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
struct aic23 *aic23 = snd_soc_component_get_drvdata(component);
/* deactivate */
- if (!snd_soc_component_is_active(component)) {
+ if (!snd_soc_component_active(component)) {
udelay(50);
snd_soc_component_write(component, TLV320AIC23_ACTIVE, 0x0);
}
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 808654b10deb..d905e03aaec7 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -449,7 +449,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol,
if (dac33->fifo_mode == ucontrol->value.enumerated.item[0])
return 0;
/* Do not allow changes while stream is running*/
- if (snd_soc_component_is_active(component))
+ if (snd_soc_component_active(component))
return -EPERM;
if (ucontrol->value.enumerated.item[0] >= DAC33_FIFO_LAST_MODE)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 26b2ee428aee..89f2bfeeb70e 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -110,7 +110,7 @@ static int uda1380_write(struct snd_soc_component *component, unsigned int reg,
/* the interpolator & decimator regs must only be written when the
* codec DAI is active.
*/
- if (!snd_soc_component_is_active(component) && (reg >= UDA1380_MVOL))
+ if (!snd_soc_component_active(component) && (reg >= UDA1380_MVOL))
return 0;
pr_debug("uda1380: hw write %x val %x\n", reg, value);
if (i2c_master_send(uda1380->i2c, data, 3) == 3) {
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index b30bfcd6a125..c56b9329240f 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -183,7 +183,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
return 0;
/* Do not allow changes while stream is running */
- if (snd_soc_component_is_active(component))
+ if (snd_soc_component_active(component))
return -EPERM;
if (ucontrol->value.enumerated.item[0] >= ARRAY_SIZE(wl1273_audio_route))
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 8036b18fdeb9..5ad905dd78b7 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -198,7 +198,7 @@ static void wm8711_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
/* deactivate */
- if (!snd_soc_component_is_active(component)) {
+ if (!snd_soc_component_active(component)) {
udelay(50);
snd_soc_component_write(component, WM8711_ACTIVE, 0x0);
}
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 95a12718f3af..8753c55c73fa 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -241,7 +241,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
if (wm8753->dai_func == ucontrol->value.enumerated.item[0])
return 0;
- if (snd_soc_component_is_active(component))
+ if (snd_soc_component_active(component))
return -EBUSY;
ioctl = snd_soc_component_read32(component, WM8753_IOCTL);
@@ -1304,7 +1304,7 @@ 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 (!snd_soc_component_is_active(component))
+ if (!snd_soc_component_active(component))
snd_soc_component_write(component, WM8753_DAC, mute_reg | 0x8);
} else {
if (mute)
--
2.17.1
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