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From: Tony Lindgren <tony@atomide.com>
To: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Mark Brown <broonie@kernel.org>,
	Liam Girdwood <lgirdwood@gmail.com>,
	Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.com>,
	alsa-devel@alsa-project.org, linux-kernel@vger.kernel.org,
	linux-omap@vger.kernel.org, "Arthur D ." <spinal.by@gmail.com>,
	Merlijn Wajer <merlijn@wizzup.org>, Pavel Machek <pavel@ucw.cz>,
	Sebastian Reichel <sre@kernel.org>,
	Jarkko Nikula <jarkko.nikula@bitmer.com>
Subject: Re: [PATCH] ASoC: cpcap: Implement set_tdm_slot for voice call support
Date: Wed, 12 Feb 2020 06:46:20 -0800	[thread overview]
Message-ID: <20200212144620.GJ64767@atomide.com> (raw)
In-Reply-To: <ae2b7d9e-d05e-54ac-4f18-27cc8c4e81a0@ti.com>

* 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 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

  reply	other threads:[~2020-02-12 14:46 UTC|newest]

Thread overview: 14+ messages / expand[flat|nested]  mbox.gz  Atom feed  top
2020-02-11 18:10 Tony Lindgren
2020-02-12  9:17 ` Peter Ujfalusi
2020-02-12 14:46   ` Tony Lindgren [this message]
2020-02-14 13:29     ` Peter Ujfalusi
2020-02-17 23:23       ` Tony Lindgren
2020-02-18 15:15         ` Peter Ujfalusi
2020-02-18 15:32           ` Tony Lindgren
2020-02-18 16:44             ` Mark Brown
2020-02-18 17:06             ` Sebastian Reichel
2020-02-18 17:42               ` Mark Brown
2020-02-19 17:39                 ` Tony Lindgren
2020-02-19 17:46                   ` Mark Brown
2020-02-19 18:49                     ` Tony Lindgren
2020-02-19 18:53               ` Tony Lindgren

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