From patchwork Thu Jul 9 01:56:25 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Kuninori Morimoto X-Patchwork-Id: 11653173 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 7853413B1 for ; Thu, 9 Jul 2020 02:01:58 +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 0CB0F206F6 for ; Thu, 9 Jul 2020 02:01:58 +0000 (UTC) Authentication-Results: mail.kernel.org; dkim=pass (1024-bit key) header.d=alsa-project.org header.i=@alsa-project.org header.b="IH+IGraI" DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org 0CB0F206F6 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 85DC415E5; Thu, 9 Jul 2020 04:01:09 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa0.perex.cz 85DC415E5 DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/simple; d=alsa-project.org; s=default; t=1594260116; bh=CLFnWNMxwvspL1ojrJkHgN2MxOvjP1069jMtlaFGyTY=; h=Date:From:Subject:To:In-Reply-To:References:Cc:List-Id: List-Unsubscribe:List-Archive:List-Post:List-Help:List-Subscribe: From; b=IH+IGraIxsFcpPU7dfxRiRZ9eVwlaCqhYOoFrrZDtfpDca6oUru/us6sVKM7TONJ1 E/TFKCH7AWypICnCjUVbk5Zu7SJ4PlvciOis+yFt+ODaRvUXrEHrdAPji1Jzn67XL7 4MHAtJlUrvKHzvJEahsWIpDpfO5gN6E9GrNgsVZI= Received: from alsa1.perex.cz (localhost.localdomain [127.0.0.1]) by alsa1.perex.cz (Postfix) with ESMTP id 8829AF8031A; Thu, 9 Jul 2020 03:56:31 +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 11DC6F80316; Thu, 9 Jul 2020 03:56:30 +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 150AFF802FE for ; Thu, 9 Jul 2020 03:56:25 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa1.perex.cz 150AFF802FE Date: 09 Jul 2020 10:56:25 +0900 X-IronPort-AV: E=Sophos;i="5.75,329,1589209200"; d="scan'208";a="51457219" Received: from unknown (HELO relmlir6.idc.renesas.com) ([10.200.68.152]) by relmlie6.idc.renesas.com with ESMTP; 09 Jul 2020 10:56:25 +0900 Received: from mercury.renesas.com (unknown [10.166.252.133]) by relmlir6.idc.renesas.com (Postfix) with ESMTP id 903F840E4BF6; Thu, 9 Jul 2020 10:56:25 +0900 (JST) Message-ID: <874kqhxxhz.wl-kuninori.morimoto.gx@renesas.com> From: Kuninori Morimoto Subject: [PATCH v3 09/21] 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: <87h7uhxxk6.wl-kuninori.morimoto.gx@renesas.com> References: <87h7uhxxk6.wl-kuninori.morimoto.gx@renesas.com> MIME-Version: 1.0 (generated by SEMI-EPG 1.14.7 - "Harue") Cc: Linux-ALSA 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 Reviewed-by: Peter Ujfalusi --- sound/soc/codecs/tlv320aic23.c | 5 +++-- sound/soc/codecs/tlv320aic26.c | 5 +++-- sound/soc/codecs/tlv320aic31xx.c | 6 ++++-- sound/soc/codecs/tlv320aic32x4.c | 5 +++-- sound/soc/codecs/tlv320aic3x.c | 5 +++-- 5 files changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index d22f75e8fb6a..68a78f0da402 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -404,7 +404,7 @@ 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; @@ -512,9 +512,10 @@ 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, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tlv320aic23_dai = { diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 032b39735643..35a4d6cf936d 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -134,7 +134,7 @@ 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); @@ -211,9 +211,10 @@ 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, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver aic26_dai = { diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 31daa60695bd..aee23b1abe6a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -972,7 +972,8 @@ 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; @@ -1378,7 +1379,8 @@ 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, + .no_capture_mute = 1, }; 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..467802875c13 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -812,7 +812,7 @@ 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; @@ -866,9 +866,10 @@ 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, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver aic32x4_dai = { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6860743ecdca..6d066bc58ac8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1216,7 +1216,7 @@ 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; @@ -1481,10 +1481,11 @@ 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, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver aic3x_dai = {