From mboxrd@z Thu Jan 1 00:00:00 1970 From: Mark Brown Subject: Re: [PATCH v5] ASoC:Add support for cs42l73 codec Date: Thu, 20 Oct 2011 14:58:11 +0100 Message-ID: <20111020135811.GA6100@sirena.org.uk> References: <1318952816-16983-1-git-send-email-brian.austin@cirrus.com> <1319013770.23438.184.camel@vkoul-udesk3> 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 0F92D247B7 for ; Thu, 20 Oct 2011 15:58:18 +0200 (CEST) Content-Disposition: inline In-Reply-To: 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: "Austin, Brian" Cc: Vinod Koul , "" , "" , "" , "" , "jeeja.kp" , "" List-Id: alsa-devel@alsa-project.org On Thu, Oct 20, 2011 at 01:54:06PM +0000, Austin, Brian wrote: > The mixer in the codec is configured by the Attenuation mixer controls. > You then select the mono/stereo output with the xSP Output Mixer Select. > I didn't think we needed to represent that in the DAPM map. That would be *extremely* surprising given that DAPM is supposed to understand the audio routing through the CODEC so it can tell where the connected audio paths go. Doesn't your current design cause too much of the device to be powered up?