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Message-ID: <1530014920.2900.50.camel@baylibre.com>
Date: Tue, 26 Jun 2018 14:08:40 +0200
From: Jerome Brunet <jbrunet@...libre.com>
To: Mark Brown <broonie@...nel.org>
Cc: jiada_wang@...tor.com, lgirdwood@...il.com, perex@...ex.cz,
tiwai@...e.com, alsa-devel@...a-project.org,
linux-kernel@...r.kernel.org
Subject: Re: [alsa-devel] [PATCH 1/1] ASoC: soc-pcm: DPCM cares BE channel
constraint
On Tue, 2018-06-26 at 12:55 +0100, Mark Brown wrote:
> On Tue, Jun 26, 2018 at 12:18:38PM +0200, Jerome Brunet wrote:
> > On Wed, 2018-06-20 at 18:25 +0900, jiada_wang@...tor.com wrote:
> > > + /* DPCM used FE & BE merged channel */
> > > + unsigned int dpcm_merged_chan:1;
> > Jiada, Mark,
> > Do you think we could extend this flag to let the link choose whether the merge
> > should be performed on the codec dais (as done here) or on the backend cpu dais
> > ?
> > I have more less the same need as Jiada but since my card uses multicodec links,
> > merging on the codec dais does not work for me.
> > Like in soc_pcm_init_runtime_hw(), we can't enforce channels min/max based on
> > the codec when there is multiple codecs on the link.
>
> Ugh, probably that'd work. The ideal thing would be to remove DPCM but
> we're stuck with it for the time being :(
>From comment here and in other mails, I get that you are not big fan of DPCM :)
It is indeed a complex beast ...
It there anything else we can use to represent audio routing within the SoC ?
The SoC I'm working on make heavy use of this, unfortunately.
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