From mboxrd@z Thu Jan 1 00:00:00 1970 From: Mark Brown Subject: Re: [PATCH 4/4] ASoC: tlv320dac33: Support for turning off the codec in BIAS_STANDBY Date: Fri, 30 Apr 2010 10:56:37 +0100 Message-ID: <20100430095636.GC21607@sirena.org.uk> References: <1272612715-13387-1-git-send-email-peter.ujfalusi@nokia.com> <1272612715-13387-5-git-send-email-peter.ujfalusi@nokia.com> <20100430084100.GB28241@rakim.wolfsonmicro.main> <201004301245.38913.peter.ujfalusi@nokia.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from cassiel.sirena.org.uk (cassiel.sirena.org.uk [80.68.93.111]) by alsa0.perex.cz (Postfix) with ESMTP id A2787243FB for ; Fri, 30 Apr 2010 11:56:37 +0200 (CEST) Content-Disposition: inline In-Reply-To: <201004301245.38913.peter.ujfalusi@nokia.com> 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: Peter Ujfalusi Cc: alsa-devel@alsa-project.org, "lrg@slimlogic.co.uk" List-Id: alsa-devel@alsa-project.org On Fri, Apr 30, 2010 at 12:45:38PM +0300, Peter Ujfalusi wrote: > On Friday 30 April 2010 11:41:00 ext Mark Brown wrote: > > Moving the playback related configuration into events on the DAC widgets > > (or probably a supply connected to the DAC widgets) seems like a good > > move for a lot of these scenarios? The core will then take care of > > ensuring that the startup sequence for the playback is called for you > > and the states can do what they're supposed to more directly. > This is not working. > Actually it works, if we come from BIAS_OFF, but... > If I restart the playback fast (within asoc timeout for BIAS change), than the > widget will not get event (since it is still powered). This means that I can not > do the needed reconfiguration for the tlv320dac33 -> audio breaks. > I will keep the current logic, but move it a bit with the idle_bias_off change. Hrm, you need to do this any time playback is started? Then just use the hooks in the normal audio stream bringup/teardown surely? It's possible that I'm missing something as a result of your list of use cases but I'd expect this to flow fairly naturally from the normal call flow.