From patchwork Thu Jul 9 01:56:20 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Kuninori Morimoto X-Patchwork-Id: 11653171 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 195EF739 for ; Thu, 9 Jul 2020 02:01:11 +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 A29CB206F6 for ; Thu, 9 Jul 2020 02:01:10 +0000 (UTC) Authentication-Results: mail.kernel.org; dkim=pass (1024-bit key) header.d=alsa-project.org header.i=@alsa-project.org header.b="sjwRdGYO" DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org A29CB206F6 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 18E729F6; Thu, 9 Jul 2020 04:00:22 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa0.perex.cz 18E729F6 DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/simple; d=alsa-project.org; s=default; t=1594260069; bh=j30rX7okzfpMvQwk9Q+T0jbmFtVo3qd81Tmi9mf1igQ=; h=Date:From:Subject:To:In-Reply-To:References:Cc:List-Id: List-Unsubscribe:List-Archive:List-Post:List-Help:List-Subscribe: From; b=sjwRdGYOzWj40SA9zsUa1S2ydZ4mKzcV0JiDhkM4V1H5EuAY8pngnQ71rlCMVVIyJ tfVfFSBK3mCExgCxyU68DRPFFRhOUPuWon67o3AaWMJ4bhRW6NVZtsFv2zN73RE6qx Ts5W/7d+IsqUvifEseG6wMn5PbuNt9eMGMBjXVxY= Received: from alsa1.perex.cz (localhost.localdomain [127.0.0.1]) by alsa1.perex.cz (Postfix) with ESMTP id 35F17F80303; Thu, 9 Jul 2020 03:56:28 +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 1BD40F80303; Thu, 9 Jul 2020 03:56:26 +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 2F016F80117 for ; Thu, 9 Jul 2020 03:56:20 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa1.perex.cz 2F016F80117 Date: 09 Jul 2020 10:56:20 +0900 X-IronPort-AV: E=Sophos;i="5.75,329,1589209200"; d="scan'208";a="51457215" Received: from unknown (HELO relmlir6.idc.renesas.com) ([10.200.68.152]) by relmlie6.idc.renesas.com with ESMTP; 09 Jul 2020 10:56:20 +0900 Received: from mercury.renesas.com (unknown [10.166.252.133]) by relmlir6.idc.renesas.com (Postfix) with ESMTP id A9DA94138B2C; Thu, 9 Jul 2020 10:56:20 +0900 (JST) Message-ID: <875zaxxxi4.wl-kuninori.morimoto.gx@renesas.com> From: Kuninori Morimoto Subject: [PATCH v3 08/21] ASoC: codecs: 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/88pm860x-codec.c | 8 +++++--- sound/soc/codecs/ad193x.c | 5 +++-- sound/soc/codecs/adau1701.c | 5 +++-- sound/soc/codecs/cpcap.c | 11 +++++++---- sound/soc/codecs/cq93vc.c | 5 +++-- sound/soc/codecs/isabelle.c | 15 +++++++++------ sound/soc/codecs/jz4770.c | 5 +++-- sound/soc/codecs/lm49453.c | 25 +++++++++++++++---------- sound/soc/codecs/ml26124.c | 5 +++-- sound/soc/codecs/nau8822.c | 5 +++-- sound/soc/codecs/rk3328_codec.c | 5 +++-- sound/soc/codecs/sgtl5000.c | 5 +++-- sound/soc/codecs/sta529.c | 5 +++-- sound/soc/codecs/tfa9879.c | 5 +++-- sound/soc/codecs/twl6040.c | 5 +++-- sound/soc/codecs/uda134x.c | 5 +++-- 16 files changed, 72 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 068914d0ef3d..cac7e557edc8 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -902,7 +902,7 @@ static const struct snd_soc_dapm_route pm860x_dapm_routes[] = { * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute. * These bits can also be used to mute. */ -static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute) +static int pm860x_mute_stream(struct snd_soc_dai *codec_dai, int mute, int direction) { struct snd_soc_component *component = codec_dai->component; int data = 0, mask = MUTE_LEFT | MUTE_RIGHT; @@ -1136,17 +1136,19 @@ static int pm860x_set_bias_level(struct snd_soc_component *component, } static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = { - .digital_mute = pm860x_digital_mute, + .mute_stream = pm860x_mute_stream, .hw_params = pm860x_pcm_hw_params, .set_fmt = pm860x_pcm_set_dai_fmt, .set_sysclk = pm860x_set_dai_sysclk, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = { - .digital_mute = pm860x_digital_mute, + .mute_stream = pm860x_mute_stream, .hw_params = pm860x_i2s_hw_params, .set_fmt = pm860x_i2s_set_dai_fmt, .set_sysclk = pm860x_set_dai_sysclk, + .no_capture_mute = 1, }; #define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 980e024a5720..f37ab7eda615 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -143,7 +143,7 @@ static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) * DAI ops entries */ -static int ad193x_mute(struct snd_soc_dai *dai, int mute) +static int ad193x_mute(struct snd_soc_dai *dai, int mute, int direction) { struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(dai->component); @@ -371,10 +371,11 @@ static int ad193x_startup(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops ad193x_dai_ops = { .startup = ad193x_startup, .hw_params = ad193x_hw_params, - .digital_mute = ad193x_mute, + .mute_stream = ad193x_mute, .set_tdm_slot = ad193x_set_tdm_slot, .set_sysclk = ad193x_set_dai_sysclk, .set_fmt = ad193x_set_dai_fmt, + .no_capture_mute = 1, }; /* codec DAI instance */ diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 115e296b2ad6..68130eaa64a4 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -573,7 +573,7 @@ static int adau1701_set_bias_level(struct snd_soc_component *component, return 0; } -static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) +static int adau1701_mute_stream(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; unsigned int mask = ADAU1701_DSPCTRL_DAM; @@ -631,8 +631,9 @@ static int adau1701_startup(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops adau1701_dai_ops = { .set_fmt = adau1701_set_dai_fmt, .hw_params = adau1701_hw_params, - .digital_mute = adau1701_digital_mute, + .mute_stream = adau1701_mute_stream, .startup = adau1701_startup, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver adau1701_dai = { diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index d7f05b384f1f..f046987ee4cd 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1216,7 +1216,7 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, return regmap_update_bits(cpcap->regmap, reg, mask, val); } -static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute) +static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); @@ -1237,7 +1237,8 @@ static const struct snd_soc_dai_ops cpcap_dai_hifi_ops = { .hw_params = cpcap_hifi_hw_params, .set_sysclk = cpcap_hifi_set_dai_sysclk, .set_fmt = cpcap_hifi_set_dai_fmt, - .digital_mute = cpcap_hifi_set_mute, + .mute_stream = cpcap_hifi_set_mute, + .no_capture_mute = 1, }; static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, @@ -1370,7 +1371,8 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute) +static int cpcap_voice_set_mute(struct snd_soc_dai *dai, + int mute, int direction) { struct snd_soc_component *component = dai->component; struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); @@ -1391,7 +1393,8 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = { .hw_params = cpcap_voice_hw_params, .set_sysclk = cpcap_voice_set_dai_sysclk, .set_fmt = cpcap_voice_set_dai_fmt, - .digital_mute = cpcap_voice_set_mute, + .mute_stream = cpcap_voice_set_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver cpcap_dai[] = { diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index b0cc61178a41..0aae5790222a 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -30,7 +30,7 @@ static const struct snd_kcontrol_new cq93vc_snd_controls[] = { SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), }; -static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +static int cq93vc_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u8 reg; @@ -87,8 +87,9 @@ static int cq93vc_set_bias_level(struct snd_soc_component *component, #define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) static const struct snd_soc_dai_ops cq93vc_dai_ops = { - .digital_mute = cq93vc_mute, + .mute_stream = cq93vc_mute, .set_sysclk = cq93vc_set_dai_sysclk, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver cq93vc_dai = { diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 3626f70f7768..79afced75d76 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -860,7 +860,7 @@ static const struct snd_soc_dapm_route isabelle_intercon[] = { { "LINEOUT2", NULL, "LINEOUT2 Driver" }, }; -static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute) +static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, ISABELLE_DAC1_SOFTRAMP_REG, BIT(4), (mute ? BIT(4) : 0)); @@ -868,7 +868,7 @@ static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute) return 0; } -static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute) +static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, ISABELLE_DAC2_SOFTRAMP_REG, BIT(4), (mute ? BIT(4) : 0)); @@ -876,7 +876,7 @@ static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute) return 0; } -static int isabelle_line_mute(struct snd_soc_dai *dai, int mute) +static int isabelle_line_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, ISABELLE_DAC3_SOFTRAMP_REG, BIT(4), (mute ? BIT(4) : 0)); @@ -1014,19 +1014,22 @@ static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static const struct snd_soc_dai_ops isabelle_hs_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, - .digital_mute = isabelle_hs_mute, + .mute_stream = isabelle_hs_mute, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops isabelle_hf_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, - .digital_mute = isabelle_hf_mute, + .mute_stream = isabelle_hf_mute, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops isabelle_line_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, - .digital_mute = isabelle_line_mute, + .mute_stream = isabelle_line_mute, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops isabelle_ul_dai_ops = { diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index 34775aa62402..e98f04749c11 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -264,7 +264,7 @@ static int jz4770_codec_pcm_trigger(struct snd_pcm_substream *substream, return ret; } -static int jz4770_codec_digital_mute(struct snd_soc_dai *dai, int mute) +static int jz4770_codec_mute_stream(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *codec = dai->component; struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec); @@ -753,7 +753,8 @@ static const struct snd_soc_dai_ops jz4770_codec_dai_ops = { .shutdown = jz4770_codec_shutdown, .hw_params = jz4770_codec_hw_params, .trigger = jz4770_codec_pcm_trigger, - .digital_mute = jz4770_codec_digital_mute, + .mute_stream = jz4770_codec_mute_stream, + .no_capture_mute = 1, }; #define JZ_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index f864b07cb0b8..06ab61f6f719 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1218,35 +1218,35 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, return 0; } -static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), (mute ? (BIT(1)|BIT(0)) : 0)); return 0; } -static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute) +static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2), (mute ? (BIT(3)|BIT(2)) : 0)); return 0; } -static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute) +static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4), (mute ? (BIT(5)|BIT(4)) : 0)); return 0; } -static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute) +static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(4), (mute ? BIT(4) : 0)); return 0; } -static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute) +static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute, int direction) { snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6), (mute ? (BIT(7)|BIT(6)) : 0)); @@ -1288,35 +1288,40 @@ static const struct snd_soc_dai_ops lm49453_headset_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, - .digital_mute = lm49453_hp_mute, + .mute_stream = lm49453_hp_mute, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops lm49453_speaker_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, - .digital_mute = lm49453_ls_mute, + .mute_stream = lm49453_ls_mute, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops lm49453_haptic_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, - .digital_mute = lm49453_ha_mute, + .mute_stream = lm49453_ha_mute, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops lm49453_ep_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, - .digital_mute = lm49453_ep_mute, + .mute_stream = lm49453_ep_mute, + .no_capture_mute = 1, }; static const struct snd_soc_dai_ops lm49453_lineout_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, - .digital_mute = lm49453_lo_mute, + .mute_stream = lm49453_lo_mute, + .no_capture_mute = 1, }; /* LM49453 dai structure. */ diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 55823bc95d06..70c17be455ca 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -372,7 +372,7 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ml26124_mute(struct snd_soc_dai *dai, int mute) +static int ml26124_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct ml26124_priv *priv = snd_soc_component_get_drvdata(component); @@ -492,9 +492,10 @@ static int ml26124_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops ml26124_dai_ops = { .hw_params = ml26124_hw_params, - .digital_mute = ml26124_mute, + .mute_stream = ml26124_mute, .set_fmt = ml26124_set_dai_fmt, .set_sysclk = ml26124_set_dai_sysclk, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver ml26124_dai = { diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 79928ddeb7a1..609aeeb27818 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -900,7 +900,7 @@ static int nau8822_hw_params(struct snd_pcm_substream *substream, return 0; } -static int nau8822_mute(struct snd_soc_dai *dai, int mute) +static int nau8822_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; @@ -967,10 +967,11 @@ static int nau8822_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops nau8822_dai_ops = { .hw_params = nau8822_hw_params, - .digital_mute = nau8822_mute, + .mute_stream = nau8822_mute, .set_fmt = nau8822_set_dai_fmt, .set_sysclk = nau8822_set_dai_sysclk, .set_pll = nau8822_set_pll, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver nau8822_dai = { diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 115706a55577..940a2fa933ed 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -107,7 +107,7 @@ static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute) +static int rk3328_mute_stream(struct snd_soc_dai *dai, int mute, int direction) { struct rk3328_codec_priv *rk3328 = snd_soc_component_get_drvdata(dai->component); @@ -316,9 +316,10 @@ static void rk3328_pcm_shutdown(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops rk3328_dai_ops = { .hw_params = rk3328_hw_params, .set_fmt = rk3328_set_dai_fmt, - .digital_mute = rk3328_digital_mute, + .mute_stream = rk3328_mute_stream, .startup = rk3328_pcm_startup, .shutdown = rk3328_pcm_shutdown, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver rk3328_dai[] = { diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index eb08976a7d06..4d6ff8114622 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -775,7 +775,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { }; /* mute the codec used by alsa core */ -static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +static int sgtl5000_mute_stream(struct snd_soc_dai *codec_dai, int mute, int direction) { struct snd_soc_component *component = codec_dai->component; u16 i2s_pwr = SGTL5000_I2S_IN_POWERUP; @@ -1160,9 +1160,10 @@ static int sgtl5000_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops sgtl5000_ops = { .hw_params = sgtl5000_pcm_hw_params, - .digital_mute = sgtl5000_digital_mute, + .mute_stream = sgtl5000_mute_stream, .set_fmt = sgtl5000_set_dai_fmt, .set_sysclk = sgtl5000_set_dai_sysclk, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver sgtl5000_dai = { diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 2881a0f7bb39..97b5f34027c0 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -251,7 +251,7 @@ static int sta529_hw_params(struct snd_pcm_substream *substream, return 0; } -static int sta529_mute(struct snd_soc_dai *dai, int mute) +static int sta529_mute(struct snd_soc_dai *dai, int mute, int direction) { u8 val = 0; @@ -291,7 +291,8 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) static const struct snd_soc_dai_ops sta529_dai_ops = { .hw_params = sta529_hw_params, .set_fmt = sta529_set_dai_fmt, - .digital_mute = sta529_mute, + .mute_stream = sta529_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver sta529_dai = { diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index abc114a3ae2b..3d8e8c2276f0 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -93,7 +93,7 @@ static int tfa9879_hw_params(struct snd_pcm_substream *substream, return 0; } -static int tfa9879_digital_mute(struct snd_soc_dai *dai, int mute) +static int tfa9879_mute_stream(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; @@ -251,8 +251,9 @@ static const struct regmap_config tfa9879_regmap = { static const struct snd_soc_dai_ops tfa9879_dai_ops = { .hw_params = tfa9879_hw_params, - .digital_mute = tfa9879_digital_mute, + .mute_stream = tfa9879_mute_stream, .set_fmt = tfa9879_set_fmt, + .no_capture_mute = 1, }; #define TFA9879_RATES SNDRV_PCM_RATE_8000_96000 diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index f34637afee51..b37203336c4e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -997,7 +997,7 @@ static void twl6040_mute_path(struct snd_soc_component *component, enum twl6040_ } } -static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) +static int twl6040_mute_stream(struct snd_soc_dai *dai, int mute, int direction) { switch (dai->id) { case TWL6040_DAI_LEGACY: @@ -1020,7 +1020,8 @@ static const struct snd_soc_dai_ops twl6040_dai_ops = { .hw_params = twl6040_hw_params, .prepare = twl6040_prepare, .set_sysclk = twl6040_set_dai_sysclk, - .digital_mute = twl6040_digital_mute, + .mute_stream = twl6040_mute_stream, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver twl6040_dai[] = { diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 1cc7f56912dc..bf9182cedb82 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -117,7 +117,7 @@ static inline void uda134x_reset(struct snd_soc_component *component) regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, 0); } -static int uda134x_mute(struct snd_soc_dai *dai, int mute) +static int uda134x_mute(struct snd_soc_dai *dai, int mute, int direction) { struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(dai->component); unsigned int mask = 1<<2; @@ -416,9 +416,10 @@ static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, .hw_params = uda134x_hw_params, - .digital_mute = uda134x_mute, + .mute_stream = uda134x_mute, .set_sysclk = uda134x_set_dai_sysclk, .set_fmt = uda134x_set_dai_fmt, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver uda134x_dai = {