From mboxrd@z Thu Jan 1 00:00:00 1970 From: Stas Sergeev Subject: Re: [PATCH] ASoC: define playback and capture streams in dummy codec Date: Fri, 12 Apr 2013 18:19:43 +0400 Message-ID: <20130412181943.0338083b@stas> References: <5167E7C4.7080501@list.ru> <20130412105943.GU9243@opensource.wolfsonmicro.com> Mime-Version: 1.0 Content-Type: multipart/mixed; boundary="MP_/tdcp.lfEuHkOwu1IxSHHVe." Return-path: Received: from smtp16.mail.ru (smtp16.mail.ru [94.100.176.153]) by alsa0.perex.cz (Postfix) with ESMTP id C682B2616B4 for ; Fri, 12 Apr 2013 16:20:05 +0200 (CEST) In-Reply-To: <20130412105943.GU9243@opensource.wolfsonmicro.com> 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 To: Mark Brown Cc: ALSA devel , Lars-Peter Clausen , Liam Girdwood , "Gabriel M. Beddingfield" , stsp@users.sourceforge.net List-Id: alsa-devel@alsa-project.org --MP_/tdcp.lfEuHkOwu1IxSHHVe. Content-Type: text/plain; charset=US-ASCII Content-Transfer-Encoding: 7bit Content-Disposition: inline On Fri, 12 Apr 2013 11:59:44 +0100 Mark Brown wrote: > > This patch adds a playback and capture streams to the dummy codec DAI > > configuration. Most permissive set of sampling rates and formats is used. > The changelog ought to explain why this is required... I can only tell that without this patch I am not getting any sound at all... which is probably a rather poor statement for the changelog. Could you please correct the changelog with the better knowledge of the details? > > Signed-off-by: Stas Sergeev > Please send patches as documented in SubmittingPatches - even as an > attachment this isn't coming out as plain text for me. OK, now it should be as a plain text. --MP_/tdcp.lfEuHkOwu1IxSHHVe. Content-Type: text/x-patch Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename=0001-ASoC-define-playback-and-capture-streams-in-dummy-co.patch >>From 22813b2124d81d8de5cde19a4e2f9490bd23baab Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Thu, 11 Apr 2013 19:07:20 +0400 Subject: [PATCH] ASoC: define playback and capture streams in dummy codec This patch adds a playback and capture streams to the dummy codec DAI configuration. Most permissive set of sampling rates and formats is used. Signed-off-by: Stas Sergeev --- sound/soc/soc-utils.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index fe4541d..bf1f92a 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -90,8 +90,33 @@ static struct snd_soc_platform_driver dummy_platform = { }; static struct snd_soc_codec_driver dummy_codec; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, }; static int snd_soc_dummy_probe(struct platform_device *pdev) -- 1.7.11.7 --MP_/tdcp.lfEuHkOwu1IxSHHVe. Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline --MP_/tdcp.lfEuHkOwu1IxSHHVe.--