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. > 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. Need to check how things are wired and how they expected to work ;) >>> + >>> + 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. But if you have a voice call initiated should not the rate be set by the set_sysclk()? >>> + >>> + 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? 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. > > Regards, > > Tony > - Péter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel