[alsa-devel] [PATCH] ASoC: cpcap: Implement set_tdm_slot for voice call support

[Tony Lindgren]

* Peter Ujfalusi <peter.ujfalusi at ti.com> [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