From patchwork Tue Jun 23 01:20:14 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Kuninori Morimoto X-Patchwork-Id: 11622539 Return-Path: Received: from mail.kernel.org (pdx-korg-mail-1.web.codeaurora.org [172.30.200.123]) by pdx-korg-patchwork-2.web.codeaurora.org (Postfix) with ESMTP id 85B4D912 for ; Wed, 24 Jun 2020 07:59:23 +0000 (UTC) Received: from alsa0.perex.cz (alsa0.perex.cz [77.48.224.243]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by mail.kernel.org (Postfix) with ESMTPS id 19D0B20DD4 for ; Wed, 24 Jun 2020 07:59:23 +0000 (UTC) Authentication-Results: mail.kernel.org; dkim=pass (1024-bit key) header.d=alsa-project.org header.i=@alsa-project.org header.b="WDl/WQa7" DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org 19D0B20DD4 Authentication-Results: mail.kernel.org; dmarc=none (p=none dis=none) header.from=renesas.com Authentication-Results: mail.kernel.org; spf=pass smtp.mailfrom=alsa-devel-bounces@alsa-project.org Received: from alsa1.perex.cz (alsa1.perex.cz [207.180.221.201]) (using TLSv1.2 with cipher AECDH-AES256-SHA (256/256 bits)) (No client certificate requested) by alsa0.perex.cz (Postfix) with ESMTPS id 85BAA17FE; Wed, 24 Jun 2020 09:58:34 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa0.perex.cz 85BAA17FE DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/simple; d=alsa-project.org; s=default; t=1592985561; bh=4SwFetcor+Y6AMDJCc25LqWZ69zlupe/NwrNSS33Nq0=; h=Date:From:Subject:To:In-Reply-To:References:Cc:List-Id: List-Unsubscribe:List-Archive:List-Post:List-Help:List-Subscribe: From; b=WDl/WQa7k9rmhih8tDezPHgGMMzFR1kOAWylyJEhr+8p7gsy4NocRZHZ317hSYhN6 8EHqqxtiUBxC2tBRjj4bNk0qLnBPaIotUqPd0bNyneZasuIC/Wo3f2+jAwg21RJOld KW/6wDIepQhbVH7ywFSjmsSUVSTTyIhPrQTu6KI0= Received: from alsa1.perex.cz (localhost.localdomain [127.0.0.1]) by alsa1.perex.cz (Postfix) with ESMTP id 93876F80343; Wed, 24 Jun 2020 09:49:55 +0200 (CEST) X-Original-To: alsa-devel@alsa-project.org Delivered-To: alsa-devel@alsa-project.org Received: by alsa1.perex.cz (Postfix, from userid 50401) id 729B3F80162; Tue, 23 Jun 2020 03:20:21 +0200 (CEST) X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on alsa1.perex.cz X-Spam-Level: X-Spam-Status: No, score=0.0 required=5.0 tests=SPF_HELO_NONE,SPF_PASS, URIBL_BLOCKED autolearn=disabled version=3.4.0 Received: from relmlie6.idc.renesas.com (relmlor2.renesas.com [210.160.252.172]) by alsa1.perex.cz (Postfix) with ESMTP id 75B30F8010D for ; Tue, 23 Jun 2020 03:20:14 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa1.perex.cz 75B30F8010D Date: 23 Jun 2020 10:20:14 +0900 X-IronPort-AV: E=Sophos;i="5.75,268,1589209200"; d="scan'208";a="50118916" Received: from unknown (HELO relmlir5.idc.renesas.com) ([10.200.68.151]) by relmlie6.idc.renesas.com with ESMTP; 23 Jun 2020 10:20:14 +0900 Received: from mercury.renesas.com (unknown [10.166.252.133]) by relmlir5.idc.renesas.com (Postfix) with ESMTP id 89EEC4001DC8; Tue, 23 Jun 2020 10:20:13 +0900 (JST) Message-ID: <874kr237e8.wl-kuninori.morimoto.gx@renesas.com> From: Kuninori Morimoto Subject: [PATCH 08/19] ASoC: codecs: tas*: merge .digital_mute() into .mute_stream() User-Agent: Wanderlust/2.15.9 Emacs/26.3 Mule/6.0 To: Mark Brown In-Reply-To: <87ftam37ko.wl-kuninori.morimoto.gx@renesas.com> References: <87ftam37ko.wl-kuninori.morimoto.gx@renesas.com> MIME-Version: 1.0 (generated by SEMI-EPG 1.14.7 - "Harue") X-Mailman-Approved-At: Wed, 24 Jun 2020 09:49:29 +0200 Cc: Shengjiu Wang , Linux-ALSA , Michael Walle , =?iso-8859-1?q?=22Heiko_St=FCbner=22?= , Neil Armstrong , David Airlie , =?iso-8859-2?q?=22Micha=B3_Miros=B3aw=22?= , Jonghwan Choi , Alexandre Belloni , Paul Cercueil , Andrzej Hajda , Frank Shi , Laurent Pinchart , Benjamin Gaignard , "Andrew F. Davis" , Fabio Estevam , Jerome Brunet , Nikita Yushchenko , Pierre-Louis Bossart , Lars-Peter Clausen , Joonyoung Shim , Matthias Reichl , Katsuhiro Suzuki , Kevin Hilman , Kai Vehmanen , Takashi Iwai , YueHaibing , Russell King , Krzysztof Kozlowski , Daniel Drake , Tzung-Bi Shih , Ludovic Desroches , Kukjin Kim , Ranjani Sridharan , Dinghao Liu , Codrin Ciubotariu , Cheng-Yi Chiang , Chun-Kuang Hu , Bartosz Golaszewski , Charles Keepax , Philipp Zabel , Jonas Karlman , Liam Girdwood , Nicolas Ferre , Chuhong Yuan , Robin Murphy , James Schulman , Inki Dae , Masahiro Yamada , Christophe JAILLET , Dan Murphy , Matthias Brugger , =?iso-8859-1?q?=22Nuno_S=E1=22?= , Vincent Abriou , Peter Ujfalusi , Jernej Skrabec , Support Opensource , Marek Szyprowski , Jason Yan , Stephen Boyd , Pankaj Bharadiya , David Rhodes , Seung-Woo Kim , Sandy Huang , Pavel Dobias , Philipp Puschmann , Kyungmin Park , Vishwas A Deshpande , Daniel Vetter , Colin Ian King , Kevin Cernekee , Lucas Stach , Shawn Guo , Peter Rosin , M R Swami Reddy X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" From: Kuninori Morimoto snd_soc_dai_digital_mute() is internally using both mute_stream() (1) or digital_mute() (2), but the difference between these 2 are only handling direction. We can merge digital_mute() into mute_stream int snd_soc_dai_digital_mute(xxx, int direction) { ... else if (dai->driver->ops->mute_stream) (1) return dai->driver->ops->mute_stream(xxx, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && dai->driver->ops->digital_mute) (2) return dai->driver->ops->digital_mute(xxx); ... } Signed-off-by: Kuninori Morimoto --- sound/soc/codecs/tas2552.c | 7 +++++-- sound/soc/codecs/tas2562.c | 7 +++++-- sound/soc/codecs/tas2770.c | 7 +++++-- sound/soc/codecs/tas571x.c | 7 +++++-- sound/soc/codecs/tas5720.c | 7 +++++-- sound/soc/codecs/tas6424.c | 7 +++++-- 6 files changed, 30 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 529c0fb93f9b..32610af4d5e7 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -465,11 +465,14 @@ static int tas2552_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas2552_mute(struct snd_soc_dai *dai, int mute) +static int tas2552_mute(struct snd_soc_dai *dai, int mute, int direction) { u8 cfg1_reg = 0; struct snd_soc_component *component = dai->component; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + if (mute) cfg1_reg |= TAS2552_MUTE; @@ -519,7 +522,7 @@ static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .set_sysclk = tas2552_set_dai_sysclk, .set_fmt = tas2552_set_dai_fmt, .set_tdm_slot = tas2552_set_dai_tdm_slot, - .digital_mute = tas2552_mute, + .mute_stream = tas2552_mute, }; /* Formats supported by TAS2552 driver. */ diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 7fae88655a0f..c818be9536be 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -334,10 +334,13 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int tas2562_mute(struct snd_soc_dai *dai, int mute) +static int tas2562_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + return snd_soc_component_update_bits(component, TAS2562_PWR_CTRL, TAS2562_MODE_MASK, mute ? TAS2562_MUTE : 0); @@ -552,7 +555,7 @@ static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { .hw_params = tas2562_hw_params, .set_fmt = tas2562_set_dai_fmt, .set_tdm_slot = tas2562_set_dai_tdm_slot, - .digital_mute = tas2562_mute, + .mute_stream = tas2562_mute, }; static struct snd_soc_dai_driver tas2562_dai[] = { diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..60ef721fb456 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -189,11 +189,14 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = { {"VSENSE", "Switch", "VMON"}, }; -static int tas2770_mute(struct snd_soc_dai *dai, int mute) +static int tas2770_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + if (mute) ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, @@ -530,7 +533,7 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, } static struct snd_soc_dai_ops tas2770_dai_ops = { - .digital_mute = tas2770_mute, + .mute_stream = tas2770_mute, .hw_params = tas2770_hw_params, .set_fmt = tas2770_set_fmt, .set_tdm_slot = tas2770_set_dai_tdm_slot, diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 5b7f9fcf6cbf..a65a874fa974 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -301,12 +301,15 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream, TAS571X_SDI_FMT_MASK, val); } -static int tas571x_mute(struct snd_soc_dai *dai, int mute) +static int tas571x_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u8 sysctl2; int ret; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0; ret = snd_soc_component_update_bits(component, @@ -354,7 +357,7 @@ static int tas571x_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops tas571x_dai_ops = { .set_fmt = tas571x_set_dai_fmt, .hw_params = tas571x_hw_params, - .digital_mute = tas571x_mute, + .mute_stream = tas571x_mute, }; diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index e159f839d928..b445f4cf035e 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -199,11 +199,14 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, return ret; } -static int tas5720_mute(struct snd_soc_dai *dai, int mute) +static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + ret = snd_soc_component_update_bits(component, TAS5720_DIGITAL_CTRL2_REG, TAS5720_MUTE, mute ? TAS5720_MUTE : 0); if (ret < 0) { @@ -604,7 +607,7 @@ static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = { .hw_params = tas5720_hw_params, .set_fmt = tas5720_set_dai_fmt, .set_tdm_slot = tas5720_set_dai_tdm_slot, - .digital_mute = tas5720_mute, + .mute_stream = tas5720_mute, }; /* diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aaba39295079..33b97a603a1d 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -252,12 +252,15 @@ static int tas6424_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas6424_mute(struct snd_soc_dai *dai, int mute) +static int tas6424_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component); unsigned int val; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + dev_dbg(component->dev, "%s() mute=%d\n", __func__, mute); if (tas6424->mute_gpio) { @@ -382,7 +385,7 @@ static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = { .hw_params = tas6424_hw_params, .set_fmt = tas6424_set_dai_fmt, .set_tdm_slot = tas6424_set_dai_tdm_slot, - .digital_mute = tas6424_mute, + .mute_stream = tas6424_mute, }; static struct snd_soc_dai_driver tas6424_dai[] = {