[alsa-devel] [PATCH 0/20] ASoC: rsnd: add Synchronous SRC and U/O error salvage support

[Mark Brown]

On Wed, Jan 14, 2015 at 01:11:11AM +0000, Kuninori Morimoto wrote:

> I asked our use case to customer Team.
> According to them, they are using this rate converter in Digital TV.
> Digital TV is including its sampling rate, and exchanges it sometimes.
> Our SoC can exchange sampling rate in SRC, but other SoC needs
> external circuit (?)

> This means my 1st explain (= 48kHz can be 48001 or 47999) seems wrong.
> They doesn't care about real audio clock.

> And, this system (= Digital TV software) is implemented as middleware.
> So, this is Renesas specific feature, can't be big task (?)

OK, that's more what I thought this was originally.  It sounds like a
fairly standard ASRC deployment where the goal is to match the rate the
system wants to play audio at to the rate that the target hardware
needs.  That's not too different to the telephony cases where the system
is in one clock domain (possibly at a very different sample rate) and
the baseband another and definitely the sort of thing that DPCM is
intended to support.

This really would be better done with DPCM with the userspace control
being provided by the driver for the tuner or the machine driver and
passed through to set the rate of the back end.  Do you think that's
achieveable (even if it's just a mostly dummy DAI here)?  That way if
this is used in a system with a CODEC with a real driver on the end of
the link things should just work.
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