From mboxrd@z Thu Jan 1 00:00:00 1970 From: Daniel Mack Subject: Re: [PATCH 2/3] ASoC: cs4270: add Master Playback Switch Date: Fri, 24 Apr 2009 15:52:16 +0200 Message-ID: <20090424135216.GF10450@buzzloop.caiaq.de> References: <1240578026-1987-1-git-send-email-daniel@caiaq.de> <1240578026-1987-2-git-send-email-daniel@caiaq.de> <20090424131738.GE26440@sirena.org.uk> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from buzzloop.caiaq.de (buzzloop.caiaq.de [212.112.241.133]) by alsa0.perex.cz (Postfix) with ESMTP id C6D4B1038A7 for ; Fri, 24 Apr 2009 15:52:21 +0200 (CEST) Content-Disposition: inline In-Reply-To: <20090424131738.GE26440@sirena.org.uk> List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Sender: alsa-devel-bounces@alsa-project.org Errors-To: alsa-devel-bounces@alsa-project.org To: Mark Brown Cc: alsa-devel@alsa-project.org, Timur Tabi List-Id: alsa-devel@alsa-project.org On Fri, Apr 24, 2009 at 02:17:38PM +0100, Mark Brown wrote: > On Fri, Apr 24, 2009 at 03:00:26PM +0200, Daniel Mack wrote: > > This new control exports cs4270's DAC MUTE capabilities. > > The DAC mute is controled by cs4270_mute function - DAC mute is handles > specially since many CODECs want to have mute synchronised with power > down of the DAC. If cs4270 doesn't need this then remove the DAI > operation too. Well, what I'm willing to implement is a way to manually mute the codec while it is running, independendly from other ways. I've seen that dai function and untruly believed that those two ways could peacefully co-exist. But as it turns out, they currently influence each other, and a manually set mute will be overridden by the next stream start via the dai function. Which is bad. I'll send a new patch for that, providing this feature in a slightly more complex fashion. However, I don't want to remove the other function. If the soc core decides to mute the codec, it should be able to do so. Daniel