From patchwork Tue Jun 23 01:20:08 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Kuninori Morimoto X-Patchwork-Id: 11622535 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 0C37192A for ; Wed, 24 Jun 2020 07:58:49 +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 9730820899 for ; Wed, 24 Jun 2020 07:58:48 +0000 (UTC) Authentication-Results: mail.kernel.org; dkim=pass (1024-bit key) header.d=alsa-project.org header.i=@alsa-project.org header.b="KxiZJ6bt" DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org 9730820899 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 34A031827; Wed, 24 Jun 2020 09:58:00 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa0.perex.cz 34A031827 DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/simple; d=alsa-project.org; s=default; t=1592985527; bh=SpO74nhNGywTDjBiqMucZBLyi0w5i3jxB3VbJRtjQbg=; h=Date:From:Subject:To:In-Reply-To:References:Cc:List-Id: List-Unsubscribe:List-Archive:List-Post:List-Help:List-Subscribe: From; b=KxiZJ6btRXwDMA4efuCsnhR4yPrmCdBEIGpgR7MkRe9c97NMSKxbGDcZ8n3McC9xX kce0AwjDAy84iqjOLb3A18gOEKrYuAHogjaDayAW1G6nfcKc2qdfXcuko2nZMS8+th 3DVdCS4hHxSAdby8rTWIEBuKdQ1VlIJDBV7lYlF4= Received: from alsa1.perex.cz (localhost.localdomain [127.0.0.1]) by alsa1.perex.cz (Postfix) with ESMTP id 4353DF8033F; Wed, 24 Jun 2020 09:49:54 +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 42E72F8015B; Tue, 23 Jun 2020 03:20:16 +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 7B50BF80157 for ; Tue, 23 Jun 2020 03:20:09 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa1.perex.cz 7B50BF80157 Date: 23 Jun 2020 10:20:08 +0900 X-IronPort-AV: E=Sophos;i="5.75,268,1589209200"; d="scan'208";a="50118900" Received: from unknown (HELO relmlir5.idc.renesas.com) ([10.200.68.151]) by relmlie6.idc.renesas.com with ESMTP; 23 Jun 2020 10:20:08 +0900 Received: from mercury.renesas.com (unknown [10.166.252.133]) by relmlir5.idc.renesas.com (Postfix) with ESMTP id B09BB4001DC8; Tue, 23 Jun 2020 10:20:07 +0900 (JST) Message-ID: <875zbi37ee.wl-kuninori.morimoto.gx@renesas.com> From: Kuninori Morimoto Subject: [PATCH 07/19] ASoC: codecs: tlv*: 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/tlv320aic23.c | 7 +++++-- sound/soc/codecs/tlv320aic26.c | 7 +++++-- sound/soc/codecs/tlv320aic31xx.c | 8 ++++++-- sound/soc/codecs/tlv320aic32x4.c | 7 +++++-- sound/soc/codecs/tlv320aic3x.c | 7 +++++-- 5 files changed, 26 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index d22f75e8fb6a..74a11ccd9ca2 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -404,11 +404,14 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, aic23->requested_adc = 0; } -static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u16 reg; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + reg = snd_soc_component_read(component, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; @@ -512,7 +515,7 @@ static const struct snd_soc_dai_ops tlv320aic23_dai_ops = { .prepare = tlv320aic23_pcm_prepare, .hw_params = tlv320aic23_hw_params, .shutdown = tlv320aic23_shutdown, - .digital_mute = tlv320aic23_mute, + .mute_stream = tlv320aic23_mute, .set_fmt = tlv320aic23_set_dai_fmt, .set_sysclk = tlv320aic23_set_dai_sysclk, }; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 032b39735643..afc2a6bf0da4 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -134,12 +134,15 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, /** * aic26_mute - Mute control to reduce noise when changing audio format */ -static int aic26_mute(struct snd_soc_dai *dai, int mute) +static int aic26_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct aic26 *aic26 = snd_soc_component_get_drvdata(component); u16 reg; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n", dai, mute); @@ -211,7 +214,7 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static const struct snd_soc_dai_ops aic26_dai_ops = { .hw_params = aic26_hw_params, - .digital_mute = aic26_mute, + .mute_stream = aic26_mute, .set_sysclk = aic26_set_sysclk, .set_fmt = aic26_set_fmt, }; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 31daa60695bd..e7f68bc46826 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -972,10 +972,14 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream, return aic31xx_setup_pll(component, params); } -static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute, + int direction) { struct snd_soc_component *component = codec_dai->component; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + if (mute) { snd_soc_component_update_bits(component, AIC31XX_DACMUTE, AIC31XX_DACMUTE_MASK, @@ -1378,7 +1382,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { .hw_params = aic31xx_hw_params, .set_sysclk = aic31xx_set_dai_sysclk, .set_fmt = aic31xx_set_dai_fmt, - .digital_mute = aic31xx_dac_mute, + .mute_stream = aic31xx_dac_mute, }; static struct snd_soc_dai_driver dac31xx_dai_driver[] = { diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 8682daec016e..19a0a02ee909 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -812,10 +812,13 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, return 0; } -static int aic32x4_mute(struct snd_soc_dai *dai, int mute) +static int aic32x4_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; + snd_soc_component_update_bits(component, AIC32X4_DACMUTE, AIC32X4_MUTEON, mute ? AIC32X4_MUTEON : 0); @@ -866,7 +869,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops aic32x4_ops = { .hw_params = aic32x4_hw_params, - .digital_mute = aic32x4_mute, + .mute_stream = aic32x4_mute, .set_fmt = aic32x4_set_dai_fmt, .set_sysclk = aic32x4_set_dai_sysclk, }; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6860743ecdca..8b3d5af987cc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1216,12 +1216,15 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, return 0; } -static int aic3x_mute(struct snd_soc_dai *dai, int mute) +static int aic3x_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u8 ldac_reg = snd_soc_component_read(component, LDAC_VOL) & ~MUTE_ON; u8 rdac_reg = snd_soc_component_read(component, RDAC_VOL) & ~MUTE_ON; + if (direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + if (mute) { snd_soc_component_write(component, LDAC_VOL, ldac_reg | MUTE_ON); snd_soc_component_write(component, RDAC_VOL, rdac_reg | MUTE_ON); @@ -1481,7 +1484,7 @@ static int aic3x_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops aic3x_dai_ops = { .hw_params = aic3x_hw_params, .prepare = aic3x_prepare, - .digital_mute = aic3x_mute, + .mute_stream = aic3x_mute, .set_sysclk = aic3x_set_dai_sysclk, .set_fmt = aic3x_set_dai_fmt, .set_tdm_slot = aic3x_set_dai_tdm_slot,