From mboxrd@z Thu Jan  1 00:00:00 1970
From: Vinod Koul <vinod.koul@intel.com>
Subject: Re: ASoC: How to mix multi input ?
Date: Mon, 25 May 2015 11:18:44 +0530
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To: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Linux-ALSA <alsa-devel@alsa-project.org>, Mark Brown <broonie@kernel.org>, Liam Girdwood <lgirdwood@gmail.com>, Simon <horms@verge.net.au>
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On Thu, May 21, 2015 at 12:20:06AM +0000, Kuninori Morimoto wrote:
> 
> Hi Mark
> 
> Thank you for your reply
> 
> >> Here, sound A/B/C will be used randomly.
> >> Does ALSA SoC already has this kind of system ?
> >
> > I *think* at least the Intel systems do this already.
> 
> Thank you. I will check it.
> 
> > > I would like to ask you how to mix multi input in ASoC.
> > > Renesas sound device(s) can mix multi input to 1 output.
> > 
> > You mean in DPCM rather than a plain CODEC?  
> (snip)
> > > codec will be started if one of A/B/C are started,
> > > and it will be stopped if all of A/B/C are stopped.
> > 
> > If you connect multiple front ends to one back end this should work
> > (with DAPM routes and so on).  If it doesn't work then that's something
> > that should work.
> 
> I considered about DPCM, but *maybe* this is not goot match for us.
> If my understanding is correct, and if I used DPCM for this,
> all of sound A/B/C are start/stop:ed in same time ?
> 
> ex)
> 	start sound => start A/B/C and codec
> 	stop  sound => stop  A/B/C and codec
> 
> I guess it works if it has this connection
> 
> 	data -> [A] -> [B] -> [C] -> codec
> 
> We would like to do is like this
> 
> 	                +-+
> 	dataA -> [A] -> |M|
> 	dataB -> [B] -> |I| -> codec
> 	dataC -> [C] -> |X|
> 	                +-+
> 
> 	init          => working:
> 	start sound A => working: codec + A
> 	start sound B => working: codec + A/B (mixed)
> 	stop  sound A => working: codec + B
> 	start sound C => working: codec + B/C (mixed)
> 	stop  sound B => working: codec + C
> 	stop  sound C => working:
> 
> I guess current DPCM doesn't support this ?
it does, you have to declare A, B, C as FE then introduce a digital mixer
(real or virtual, depending on the system) and the a BE which is your HW DAI
here and connects to codec

This works very well and you can start/stop A, B, C independently. Codec
will be on as along as any of the inputs is On.

See the sound/soc/intel/atom for reference

-- 
~Vinod