From: Mark Brown <broonie@opensource.wolfsonmicro.com>
To: "Austin, Brian" <Brian.Austin@cirrus.com>
Cc: Vinod Koul <vinod.koul@linux.intel.com>,
"<alsa-devel@alsa-project.org>" <alsa-devel@alsa-project.org>,
"<ramesh.babu@intel.com>" <ramesh.babu@intel.com>,
"<lrg@ti.com>" <lrg@ti.com>,
"<heelrod@heelrod.ad.cirrus.com>" <heelrod@heelrod.ad.cirrus.com>,
"jeeja.kp" <jeeja.kp@intel.com>,
"<joe@nucleusys.com>" <joe@nucleusys.com>
Subject: Re: [PATCH v5] ASoC:Add support for cs42l73 codec
Date: Thu, 20 Oct 2011 14:58:11 +0100 [thread overview]
Message-ID: <20111020135811.GA6100@sirena.org.uk> (raw)
In-Reply-To: <EE74D414-85CB-4382-B495-46842866C963@cirrus.com>
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?
next prev parent reply other threads:[~2011-10-20 13:58 UTC|newest]
Thread overview: 14+ messages / expand[flat|nested] mbox.gz Atom feed top
2011-10-18 15:46 [PATCH v5] ASoC:Add support for cs42l73 codec Brian Austin
2011-10-19 8:42 ` Vinod Koul
2011-10-20 13:54 ` Austin, Brian
2011-10-20 13:58 ` Mark Brown [this message]
2011-10-20 13:54 ` Vinod Koul
2011-10-20 14:11 ` Austin, Brian
2011-10-20 14:20 ` Mark Brown
2011-10-20 16:23 ` Austin, Brian
[not found] ` <4EA06916.6090601@nucleusys.com>
2011-10-20 20:00 ` Mark Brown
2011-10-20 20:26 ` Austin, Brian
2011-10-20 21:28 ` Mark Brown
2011-10-20 21:34 ` Austin, Brian
2011-10-21 13:23 ` Brian Austin
2011-10-21 16:00 ` Mark Brown
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