Hi Tony,
On 18/02/2020 1.23, Tony Lindgren wrote:
> * Peter Ujfalusi <peter.ujfalusi@xxxxxx> [200214 13:30]:
>> 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.
>
> OK yeah seems at least bluetooth would need to be also handled
> in the set_tdm_slots.
set_tdm_slots() is for setting how the TDM slots supposed to be used by
the component and not really for things to configure different operating
modes.
If you hardwire things in set_tdm_slots() for the droid4 then how the
codec driver can be reused in other setups?
>>>> 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()?
>
> Hmm does set_sysclk called from modem codec know that cpcap codec
> is the clock master based on bitclock-master and set the rate
> for cpcap codec?
Neither component should call set_sysclk, set_tdm_slots. The machine
driver should as it is the only one who know how things are wired...
>
>>>> 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.
>
> Sounds good to me.
>
> Tony
>
- Péter
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