From patchwork Thu Jul 9 01:56:30 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Kuninori Morimoto X-Patchwork-Id: 11653175 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 7FA6C739 for ; Thu, 9 Jul 2020 02:02:24 +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 1820720760 for ; Thu, 9 Jul 2020 02:02:24 +0000 (UTC) Authentication-Results: mail.kernel.org; dkim=pass (1024-bit key) header.d=alsa-project.org header.i=@alsa-project.org header.b="ljxsNE+g" DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org 1820720760 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 A274C15DC; Thu, 9 Jul 2020 04:01:35 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa0.perex.cz A274C15DC DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/simple; d=alsa-project.org; s=default; t=1594260142; bh=9rX+ttn7KsLQYmTpzgOCf+HilPET34TTOh3Gui5fI30=; h=Date:From:Subject:To:In-Reply-To:References:Cc:List-Id: List-Unsubscribe:List-Archive:List-Post:List-Help:List-Subscribe: From; b=ljxsNE+gl3+QjlkhsJOx4NfrtNnLCSxBDUOnk6SzV8d7fKx405+Dm9PrU9kIaLIgF ZCI8bWEIGTT41N+KVoJvnyX0wvZx2UQc9pw+gsCcXfqEcDuZPuC0MwvS5ySwj++GKC kOQgu2ZVhJFItRr4ILEnsWmbjUfycdE8nltHxJ8M= Received: from alsa1.perex.cz (localhost.localdomain [127.0.0.1]) by alsa1.perex.cz (Postfix) with ESMTP id 8A27CF80329; Thu, 9 Jul 2020 03:56:38 +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 46378F8032A; Thu, 9 Jul 2020 03:56:37 +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 C4F4EF80322 for ; Thu, 9 Jul 2020 03:56:30 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa1.perex.cz C4F4EF80322 Date: 09 Jul 2020 10:56:30 +0900 X-IronPort-AV: E=Sophos;i="5.75,329,1589209200"; d="scan'208";a="51457228" Received: from unknown (HELO relmlir5.idc.renesas.com) ([10.200.68.151]) by relmlie6.idc.renesas.com with ESMTP; 09 Jul 2020 10:56:30 +0900 Received: from mercury.renesas.com (unknown [10.166.252.133]) by relmlir5.idc.renesas.com (Postfix) with ESMTP id 4AD72400A8A7; Thu, 9 Jul 2020 10:56:30 +0900 (JST) Message-ID: <873661xxhu.wl-kuninori.morimoto.gx@renesas.com> From: Kuninori Morimoto Subject: [PATCH v3 10/21] 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: <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/tas2552.c | 5 +++-- sound/soc/codecs/tas2562.c | 5 +++-- sound/soc/codecs/tas2770.c | 5 +++-- sound/soc/codecs/tas571x.c | 5 +++-- sound/soc/codecs/tas5720.c | 5 +++-- sound/soc/codecs/tas6424.c | 5 +++-- 6 files changed, 18 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 529c0fb93f9b..3a153526e47f 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -465,7 +465,7 @@ 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; @@ -519,7 +519,8 @@ 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, + .no_capture_mute = 1, }; /* Formats supported by TAS2552 driver. */ diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 5c28af370bd4..e74628061040 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -394,7 +394,7 @@ 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; @@ -612,7 +612,8 @@ 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, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas2562_dai[] = { diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..4538b2d0216f 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -189,7 +189,7 @@ 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; @@ -530,10 +530,11 @@ 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, + .no_capture_mute = 1, }; #define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 5b7f9fcf6cbf..835a723ce5bc 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -301,7 +301,7 @@ 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; @@ -354,7 +354,8 @@ 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, + .no_capture_mute = 1, }; diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index e159f839d928..139ac5e683bf 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -199,7 +199,7 @@ 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; @@ -604,7 +604,8 @@ 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, + .no_capture_mute = 1, }; /* diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aaba39295079..6198138e693a 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -252,7 +252,7 @@ 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); @@ -382,7 +382,8 @@ 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, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas6424_dai[] = {