>
> > > For speakers and headsets its 48Khz, 2 ch and 24 bit and this
> > > setting is done based on the front-end dai For speakers, however
> > > support only
> > > 16 bit, so we set it back to 16 bit If the front end dai is dmic,
> > > then the channel
> > is set to either 2 or 4 dmic_constraints. No other formats need to be set.
> > >
> > > All the SSP1 usages do not have the same parameters (as dmic is on
> > > SSP1 and its different as given above) Most parameters are same for
> > speakers and headset which are on different SSP. This is the reason we
> > had a single fixup function.
> On SSP1, for dmic we need to fix the channels which is derived from dmic_num
> of the snd_soc_acpi_mach structure based on the number of dmic on the
> board.
> The channel is something that might be different from speakers.
> We might not want to constraint the dmic capture to always be 48Khz as well.
> Given this, there seems to me, 2 ways to set it:
> 1. Derive if the fixup is being called for dmic or speaker 2. Having a new dailink
>
> If #2 is not preferred (going by Pierre's comments), can we use rtd-
> >cpu_dai/codec_dai->name to figure out if its for dmic or speaker?
> I can test this and get back to you.
Tested and the following is something we can use without creating a new dailink.
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
if (!strcmp(codec_dai->name, KBL_REALTEK_DMIC_CODEC_DAI)) {
if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
channels->min = channels->max = 2;
else
channels->min = channels->max = 4;
} else {
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
snd_mask_none(fmt);
snd_mask_set_format(fmt, pcm_fmt);
}
Pierre, thoughts?
