ASoC: cpcap: Remove now redundant non_legacy_dai_naming flag

The ASoC core has now been changed to default to the non-legacy DAI
naming, as such drivers using the new scheme no longer need to specify
the non_legacy_dai_naming flag.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220623125250.2355471-73-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: cpcap: Check for NULL pointer after calling of_get_child_by_name

If the device does not exist, of_get_child_by_name() will return NULL
pointer.
And devm_snd_soc_register_component() does not check it.
Also, I have noticed that cpcap_codec_driver has not been used yet.
Therefore, it should be better to check it in order to avoid the future
dereference of the NULL pointer.

Fixes: f6cdf2d3445d ("ASoC: cpcap: new codec")
Signed-off-by: Jiasheng Jiang <jiasheng@iscas.ac.cn>
Link: https://lore.kernel.org/r/20220111025048.524134-1-jiasheng@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: cpcap: Use modern ASoC DAI format terminology

As part of moving to remove the old style defines for the bus clocks update
the cpcap driver to use more modern terminology for clocking.

Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210921213034.31427-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: cpcap: fix microphone timeslot mask

The correct mask is 0x1f8 (Bit 3-8), but due to missing BIT() 0xf (Bit
0-3) was set instead. This means setting of CPCAP_BIT_MIC1_RX_TIMESLOT0
(Bit 3) still worked (part of both masks). On the other hand the code
does not properly clear the other MIC timeslot bits. I think this
is not a problem, since they are probably initialized to 0 and not
touched by the driver anywhere else. But the mask also contains some
wrong bits, that will be cleared. Bit 0 (CPCAP_BIT_SMB_CDC) should be
safe, since the driver enforces it to be 0 anyways.

Bit 1-2 are CPCAP_BIT_FS_INV and CPCAP_BIT_CLK_INV. This means enabling
audio recording forces the codec into SND_SOC_DAIFMT_NB_NF mode, which
is obviously bad.

The bug probably remained undetected, because there are not many use
cases for routing microphone to the CPU on platforms using cpcap and
user base is small. I do remember having some issues with bad sound
quality when testing voice recording back when I wrote the driver.
It probably was this bug.

Fixes: f6cdf2d3445d ("ASoC: cpcap: new codec")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Reviewed-by: Tony Lindgren <tony@atomide.com>
Link: https://lore.kernel.org/r/20210123172945.3958622-1-sre@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: cpcap: Implement set_tdm_slot for voice call support

ASoC: cpcap: Implement set_tdm_slot for voice call support

For using cpcap for voice calls, we need to route audio directly from
the modem to cpcap for TDM (Time Division Multiplexing). The voice call
is direct data between the modem and cpcap with no CPU involvment. In
this mode, the cpcap related audio mixer controls work for the speaker
selection and volume though.

To do this, we need to implement standard snd_soc_dai_set_tdm_slot()
for cpcap. Then the modem codec driver can use snd_soc_dai_set_sysclk(),
snd_soc_dai_set_fmt(), and snd_soc_dai_set_tdm_slot() to configure a
voice call.

Let's add cpcap_voice_set_tdm_slot() for this, and cpcap_voice_call()
helper to configure the additional registers needed for voice call.

Let's also clear CPCAP_REG_VAUDIOC on init in case we have the bit for
CPCAP_BIT_VAUDIO_MODE0 set on init.

Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Link: https://lore.kernel.org/r/20210112174704.GA13496@duo.ucw.cz
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: codecs: merge .digital_mute() into .mute_stream()

snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream

int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/875zaxxxi4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: cpcap: replace codec to component

Now we can replace Codec to Component. Let's do it.

Note:
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 0 -> .idle_bias_on = 1
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: cpcap: new codec

Motorola CPCAP is a PMIC with audio functionality, that can be
found on Motorola Droid 4 and probably a few other phones from
Motorola's Droid series.

The driver has been written from scratch using Motorola's Android
driver, register dumps from running Android and datasheet for NXP
MC13783UG (which is similar to Motorola CPCAP, but not the same).

The chip provides two audio interfaces, that can be muxed to two
different audio codecs. One provides support for stereo output
(named StDAC or HiFi), while the other only provides mono output
(named Voice). Only the Voice codec provides a Capture interface.

Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>