* Peter Ujfalusi <peter.ujfalusi@xxxxxx> [200214 13:30]:
> Hi Tony,
>
> On 12/02/2020 16.46, Tony Lindgren wrote:
> > * Peter Ujfalusi <peter.ujfalusi@xxxxxx> [200212 09:18]:
> >> On 11/02/2020 20.10, Tony Lindgren wrote:
> >>> +static int cpcap_voice_set_tdm_slot(struct snd_soc_dai *dai,
> >>> + unsigned int tx_mask, unsigned int rx_mask,
> >>> + int slots, int slot_width)
> >>> +{
> >>> + struct snd_soc_component *component = dai->component;
> >>> + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
> >>> + int err, ts_mask, mask;
> >>> + bool voice_call;
> >>> +
> >>> + /*
> >>> + * Primitive test for voice call, probably needs more checks
> >>> + * later on for 16-bit calls detected, Bluetooth headset etc.
> >>> + */
> >>> + if (tx_mask == 0 && rx_mask == 1 && slot_width == 8)
> >>> + voice_call = true;
> >>> + else
> >>> + voice_call = false;
> >>
> >> You only have voice call if only rx slot0 is in use?
> >
> > Yeah so it seems. Then there's the modem to wlcore bluetooth path that
> > I have not looked at. But presumably that's again just configuring some
> > tdm slot on the PMIC.
> >
> >> If you record mono on the voice DAI, then rx_mask is also 1, no?
> >
> > It is above :) But maybe I don't follow what you're asking here
>
> If you arecrod -Dvoice_pcm -c1 -fS8 > /dev/null
> then it is reasonable that the machine driver will set rx_mask = 1
>
> > and maybe you have some better check in mind.
>
> Not sure, but relying on set_tdm_slots to decide if we are in a call
> case does not sound right.
OK yeah seems at least bluetooth would need to be also handled
in the set_tdm_slots.
> >> You will also set the sampling rate for voice in
> >> cpcap_voice_hw_params(), but that is for normal playback/capture, right?
> >
> > Yeah so normal playback/capture is already working with cpcap codec driver
> > with mainline Linux. The voice call needs to set rate to 8000.
>
> But if you have a voice call initiated should not the rate be set by the
> set_sysclk()?
Hmm does set_sysclk called from modem codec know that cpcap codec
is the clock master based on bitclock-master and set the rate
for cpcap codec?
> >> It feels like that these should be done via DAPM with codec to codec route?
> >
> > Sure if you have some better way of doing it :) Do you have an example to
> > point me to?
>
> Something along the lines of:
> https://mailman.alsa-project.org/pipermail/alsa-devel/2020-February/162915.html
>
> The it is a matter of building and connecting the DAPM routes between
> the two codec and with a flip of the switch you would have audio flowing
> between them.
Sounds good to me.
Tony
