* 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 and
maybe you have some better check in mind.
I have no idea where we would implement recording voice calls for
example, I guess mcbsp could do that somewhere to dump out a tdm slot
specific traffic.
> > +
> > + ts_mask = 0x7 << CPCAP_BIT_MIC2_TIMESLOT0;
> > + ts_mask |= 0x7 << CPCAP_BIT_MIC1_RX_TIMESLOT0;
> > +
> > + mask = (tx_mask & 0x7) << CPCAP_BIT_MIC2_TIMESLOT0;
> > + mask |= (rx_mask & 0x7) << CPCAP_BIT_MIC1_RX_TIMESLOT0;
> > +
> > + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
> > + ts_mask, mask);
> > + if (err)
> > + return err;
> > +
> > + err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, slot_width * 1000);
> > + if (err)
> > + return err;
>
> 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.
> > +
> > + err = cpcap_voice_call(cpcap, dai, voice_call);
> > + if (err)
> > + return err;
>
> 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?
Regards,
Tony
