[v3,11/21] ASoC: codecs: use snd_soc_xxx_active()

Message ID	87wo5e58j4.wl-kuninori.morimoto.gx@renesas.com
State	Superseded
Headers	show Return-Path: <SRS0=wzGW=65=alsa-project.org=alsa-devel-bounces@kernel.org> DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org C15572065F Date: 15 May 2020 09:47:11 +0900 Message-ID: <87wo5e58j4.wl-kuninori.morimoto.gx@renesas.com> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Subject: [PATCH v3 11/21] ASoC: codecs: use snd_soc_xxx_active() User-Agent: Wanderlust/2.15.9 Emacs/25.2 Mule/6.0 MIME-Version: 1.0 (generated by SEMI-EPG 1.14.7 - "Harue") To: Mark Brown <broonie@kernel.org> In-Reply-To: <87d0766n64.wl-kuninori.morimoto.gx@renesas.com> References: <87d0766n64.wl-kuninori.morimoto.gx@renesas.com> Content-Type: text/plain; charset=US-ASCII Cc: Linux-ALSA <alsa-devel@alsa-project.org> Precedence: list Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" <alsa-devel-bounces@alsa-project.org>
Series	[v3,01/21] ASoC: soc-pcm: replace snd_soc_runtime_activate()/deactivate() to macro \| expand [v3,01/21] ASoC: soc-pcm: replace snd_soc_runtime_activate()/deactivate() to macro [v3,02/21] ASoC: soc-dai: add snd_soc_dai_action() [v3,06/21] ASoC: soc-dai: add snd_soc_dai_stream_active() [v3,08/21] ASoC: atomel: use snd_soc_xxx_active() [v3,10/21] ASoC: cirrus: use snd_soc_xxx_active() [v3,11/21] ASoC: codecs: use snd_soc_xxx_active() [v3,13/21] ASoC: intel: use snd_soc_xxx_active() [v3,15/21] ASoC: mediatek: use snd_soc_xxx_active() [v3,17/21] ASoC: pxa: use snd_soc_xxx_active() [v3,19/21] ASoC: uniphier: use snd_soc_xxx_active() [v3,21/21] ASoC: cleanup dai / component active code

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)

[v3,11/21] ASoC: codecs: use snd_soc_xxx_active()

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