* 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 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel