Hi Tony,
On 12/02/2020 16.46, Tony Lindgren wrote:
- Peter Ujfalusi peter.ujfalusi@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
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.h...
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
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