From patchwork Thu Jan 31 13:30:18 2019 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Charles Keepax X-Patchwork-Id: 10790713 Return-Path: Received: from mail.wl.linuxfoundation.org (pdx-wl-mail.web.codeaurora.org [172.30.200.125]) by pdx-korg-patchwork-2.web.codeaurora.org (Postfix) with ESMTP id CBD6517E9 for ; Thu, 31 Jan 2019 14:26:19 +0000 (UTC) Received: from mail.wl.linuxfoundation.org (localhost [127.0.0.1]) by mail.wl.linuxfoundation.org (Postfix) with ESMTP id B471C310D8 for ; Thu, 31 Jan 2019 14:26:19 +0000 (UTC) Received: by mail.wl.linuxfoundation.org (Postfix, from userid 486) id B2F89310EE; Thu, 31 Jan 2019 14:26:19 +0000 (UTC) X-Spam-Checker-Version: SpamAssassin 3.3.1 (2010-03-16) on pdx-wl-mail.web.codeaurora.org X-Spam-Level: X-Spam-Status: No, score=-2.9 required=2.0 tests=BAYES_00,MAILING_LIST_MULTI, RCVD_IN_DNSWL_NONE autolearn=ham version=3.3.1 Received: from alsa0.perex.cz (alsa0.perex.cz [77.48.224.243]) by mail.wl.linuxfoundation.org (Postfix) with ESMTP id 39264310D8 for ; Thu, 31 Jan 2019 14:26:18 +0000 (UTC) Received: from alsa0.perex.cz (localhost [127.0.0.1]) by alsa0.perex.cz (Postfix) with ESMTP id ED49326791D; Thu, 31 Jan 2019 14:30:29 +0100 (CET) X-Original-To: alsa-devel@alsa-project.org Delivered-To: alsa-devel@alsa-project.org Received: by alsa0.perex.cz (Postfix, from userid 1000) id 798E2267956; Thu, 31 Jan 2019 14:30:27 +0100 (CET) Received: from mx0b-001ae601.pphosted.com (mx0b-001ae601.pphosted.com [67.231.152.168]) by alsa0.perex.cz (Postfix) with ESMTP id DE821267448 for ; Thu, 31 Jan 2019 14:30:24 +0100 (CET) Received: from pps.filterd (m0077474.ppops.net [127.0.0.1]) by mx0b-001ae601.pphosted.com (8.16.0.27/8.16.0.27) with SMTP id x0VDU29s023546; Thu, 31 Jan 2019 07:30:22 -0600 Authentication-Results: ppops.net; spf=none smtp.mailfrom=ckeepax@opensource.cirrus.com Received: from mail3.cirrus.com ([87.246.76.56]) by mx0b-001ae601.pphosted.com with ESMTP id 2q8muvqv0m-1; Thu, 31 Jan 2019 07:30:21 -0600 Received: from EX17.ad.cirrus.com (ex17.ad.cirrus.com [172.20.9.81]) by mail3.cirrus.com (Postfix) with ESMTP id 5895E611C8AF; Thu, 31 Jan 2019 07:30:27 -0600 (CST) Received: from imbe.wolfsonmicro.main (198.61.95.81) by EX17.ad.cirrus.com (172.20.9.81) with Microsoft SMTP Server id 14.3.408.0; Thu, 31 Jan 2019 13:30:20 +0000 Received: from algalon.ad.cirrus.com (algalon.ad.cirrus.com [198.90.251.122]) by imbe.wolfsonmicro.main (8.14.4/8.14.4) with ESMTP id x0VDUJbV017675; Thu, 31 Jan 2019 13:30:19 GMT From: Charles Keepax To: Date: Thu, 31 Jan 2019 13:30:18 +0000 Message-ID: <20190131133019.1965-1-ckeepax@opensource.cirrus.com> X-Mailer: git-send-email 2.11.0 MIME-Version: 1.0 X-Proofpoint-Spam-Details: rule=notspam policy=default score=0 priorityscore=1501 malwarescore=0 suspectscore=4 phishscore=0 bulkscore=0 spamscore=0 clxscore=1015 lowpriorityscore=0 mlxscore=0 impostorscore=0 mlxlogscore=473 adultscore=0 classifier=spam adjust=0 reason=mlx scancount=1 engine=8.0.1-1810050000 definitions=main-1901310106 Cc: patches@opensource.cirrus.com, alsa-devel@alsa-project.org, lgirdwood@gmail.com Subject: [alsa-devel] [PATCH 1/2] ASoC: dapm: Only power up active channels from a DAI X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.14 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-bounces@alsa-project.org X-Virus-Scanned: ClamAV using ClamSMTP Currently all widgets attached to a DAI link will be powered up when the DAI is active, however this may include routes that are not actually in use if there are unused channels available on the DAI. The macros for creating AIF widgets already include an entry for slot, it is proposed to change that to channel. The effective difference here being respresenting the logical channel index rather than the physical slot index. The CODECs currently using the slot entry on the DAPM_AIF macros are using it in a manner consistent with this, the CODECs not using it just have the field set to zero. A variable is added to snd_soc_dapm_widget to represent this channel index and then for each AIF widget attached to a DAI this is compared against the number of channels on the stream. Enabling the links for those which will be in use. This has the nice property that the CODECs which haven't used the slot/channel entry in the macro will function exactly as before due to all the AIF widgets having a channel of zero and a stream by definition having at least one channel. Signed-off-by: Charles Keepax --- include/sound/soc-dapm.h | 22 +++++++++----- sound/soc/soc-dapm.c | 76 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 4 +++ 3 files changed, 94 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 46f2ba3ffcb7c..79b4ddfb8e9e2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -214,21 +214,21 @@ struct device; .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ -#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wchan, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } -#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } +#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wchan, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } -#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wchan, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } -#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } +#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wchan, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, \ @@ -407,6 +407,10 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); +int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); + /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); @@ -627,6 +631,8 @@ struct snd_soc_dapm_widget { int endpoints[2]; struct clk *clk; + + int channel; }; struct snd_soc_dapm_update { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c4c134195392..1714f612451e1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2525,6 +2525,78 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); +static int dapm_update_dai_chan(struct snd_soc_dapm_path *p, + struct snd_soc_dapm_widget *w, + int channels) +{ + switch (w->id) { + case snd_soc_dapm_aif_out: + case snd_soc_dapm_aif_in: + break; + default: + return 0; + } + + dev_dbg(w->dapm->dev, "%s DAI route %s -> %s\n", + w->channel < channels ? "Connecting" : "Disconnecting", + p->source->name, p->sink->name); + + if (w->channel < channels) + soc_dapm_connect_path(p, true, "dai update"); + else + soc_dapm_connect_path(p, false, "dai update"); + + return 0; +} + +static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int dir = substream->stream; + int channels = params_channels(params); + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + int ret; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + w = dai->playback_widget; + else + w = dai->capture_widget; + + dev_dbg(dai->dev, "Update DAI routes for %s %s\n", dai->name, + dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + ret = dapm_update_dai_chan(p, p->sink, channels); + if (ret < 0) + return ret; + } + + snd_soc_dapm_widget_for_each_source_path(w, p) { + ret = dapm_update_dai_chan(p, p->source, channels); + if (ret < 0) + return ret; + } + + return 0; +} + +int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret; + + mutex_lock_nested(&rtd->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = dapm_update_dai_unlocked(substream, params, dai); + mutex_unlock(&rtd->card->dapm_mutex); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai); + /* * dapm_update_widget_flags() - Re-compute widget sink and source flags * @w: The widget for which to update the flags @@ -3689,6 +3761,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; + + dapm_update_dai_unlocked(&substream, params, source); } substream.stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -3709,6 +3783,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; + + dapm_update_dai_unlocked(&substream, params, sink); } break; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 03f36e534050f..a5b40e82dea4a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -969,6 +969,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->channels = params_channels(&codec_params); codec_dai->sample_bits = snd_pcm_format_physical_width( params_format(&codec_params)); + + snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); } ret = soc_dai_hw_params(substream, params, cpu_dai); @@ -998,6 +1000,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); + snd_soc_dapm_update_dai(substream, params, cpu_dai); + ret = soc_pcm_params_symmetry(substream, params); if (ret) goto component_err;