Messages in this thread | | | Subject | Re: [alsa-devel] [PATCH 1/1] ASoC: soc-pcm: DPCM cares BE channel constraint | From | Jerome Brunet <> | Date | Tue, 26 Jun 2018 14:08:40 +0200 |
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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@mentor.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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