Re: [PATCH] ASoC: cpcap: Implement set_tdm_slot for voice call support

[Peter Ujfalusi <peter.ujfalusi@ti.com>]

On 11/02/2020 20.10, Tony Lindgren wrote:
> 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.
> 
> Cc: Arthur D. <spinal.by@gmail.com>
> Cc: Merlijn Wajer <merlijn@wizzup.org>
> Cc: Pavel Machek <pavel@ucw.cz>
> Cc: Sebastian Reichel <sre@kernel.org>
> Signed-off-by: Tony Lindgren <tony@atomide.com>
> ---
>  sound/soc/codecs/cpcap.c | 123 +++++++++++++++++++++++++++++++++++++++
>  1 file changed, 123 insertions(+)
> 
> diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
> --- a/sound/soc/codecs/cpcap.c
> +++ b/sound/soc/codecs/cpcap.c
> @@ -16,6 +16,14 @@
>  #include <sound/soc.h>
>  #include <sound/tlv.h>
>  
> +/* Register 512 CPCAP_REG_VAUDIOC --- Audio Regulator and Bias Voltage */
> +#define CPCAP_BIT_AUDIO_LOW_PWR           6
> +#define CPCAP_BIT_AUD_LOWPWR_SPEED        5
> +#define CPCAP_BIT_VAUDIOPRISTBY           4
> +#define CPCAP_BIT_VAUDIO_MODE1            2
> +#define CPCAP_BIT_VAUDIO_MODE0            1
> +#define CPCAP_BIT_V_AUDIO_EN              0
> +
>  /* Register 513 CPCAP_REG_CC     --- CODEC */
>  #define CPCAP_BIT_CDC_CLK2                15
>  #define CPCAP_BIT_CDC_CLK1                14
> @@ -221,6 +229,7 @@ struct cpcap_reg_info {
>  };
>  
>  static const struct cpcap_reg_info cpcap_default_regs[] = {
> +	{ CPCAP_REG_VAUDIOC, 0x003F, 0x0000 },
>  	{ CPCAP_REG_CC, 0xFFFF, 0x0000 },
>  	{ CPCAP_REG_CC, 0xFFFF, 0x0000 },
>  	{ CPCAP_REG_CDI, 0xBFFF, 0x0000 },
> @@ -1370,6 +1379,119 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
>  	return 0;
>  }
>  
> +/*
> + * Configure codec for voice call if requested.
> + *
> + * We can configure most with snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt()
> + * and snd_soc_dai_set_tdm_slot(). This function configures the rest of the
> + * cpcap related hardware as CPU is not involved in the voice call.
> + */
> +static int cpcap_voice_call(struct cpcap_audio *cpcap, struct snd_soc_dai *dai,
> +			    bool voice_call)
> +{
> +	int mask, err;
> +
> +	/* Modem to codec VAUDIO_MODE1 */
> +	mask = BIT(CPCAP_BIT_VAUDIO_MODE1);
> +	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_VAUDIOC,
> +				 mask, voice_call ? mask : 0);
> +	if (err)
> +		return err;
> +
> +	/* Clear MIC1_MUX for call */
> +	mask = BIT(CPCAP_BIT_MIC1_MUX);
> +	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
> +				 mask, voice_call ? 0 : mask);
> +	if (err)
> +		return err;
> +
> +	/* Set MIC2_MUX for call */
> +	mask = BIT(CPCAP_BIT_MB_ON1L) | BIT(CPCAP_BIT_MB_ON1R) |
> +		BIT(CPCAP_BIT_MIC2_MUX) | BIT(CPCAP_BIT_MIC2_PGA_EN);
> +	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
> +				 mask, voice_call ? mask : 0);
> +	if (err)
> +		return err;
> +
> +	/* Enable LDSP for call */
> +	mask = BIT(CPCAP_BIT_A2_LDSP_L_EN) | BIT(CPCAP_BIT_A2_LDSP_R_EN);
> +	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXOA,
> +				 mask, voice_call ? mask : 0);
> +	if (err)
> +		return err;
> +
> +	/* Enable CPCAP_BIT_PGA_CDC_EN for call */
> +	mask = BIT(CPCAP_BIT_PGA_CDC_EN);
> +	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA,
> +				 mask, voice_call ? mask : 0);
> +	if (err)
> +		return err;
> +
> +	/* Unmute voice for call */
> +	if (dai) {
> +		err = snd_soc_dai_digital_mute(dai, !voice_call,
> +					       SNDRV_PCM_STREAM_PLAYBACK);
> +		if (err)
> +			return err;
> +	}
> +
> +	/* Set modem to codec mic CDC and HPF for call */
> +	mask = BIT(CPCAP_BIT_MIC2_CDC_EN) | BIT(CPCAP_BIT_CDC_EN_RX) |
> +	       BIT(CPCAP_BIT_AUDOHPF_1) | BIT(CPCAP_BIT_AUDOHPF_0) |
> +	       BIT(CPCAP_BIT_AUDIHPF_1) | BIT(CPCAP_BIT_AUDIHPF_0);
> +	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CC,
> +				 mask, voice_call ? mask : 0);
> +	if (err)
> +		return err;
> +
> +	/* Enable modem to codec CDC for call*/
> +	mask = BIT(CPCAP_BIT_CDC_CLK_EN);
> +	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
> +				 mask, voice_call ? mask : 0);
> +
> +	return err;
> +}
> +
> +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?
If you record mono on the voice DAI, then rx_mask is also 1, no?

> +
> +	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?

> +
> +	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?

> +
> +	return 0;
> +}
> +
>  static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute)
>  {
>  	struct snd_soc_component *component = dai->component;
> @@ -1391,6 +1513,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = {
>  	.hw_params	= cpcap_voice_hw_params,
>  	.set_sysclk	= cpcap_voice_set_dai_sysclk,
>  	.set_fmt	= cpcap_voice_set_dai_fmt,
> +	.set_tdm_slot	= cpcap_voice_set_tdm_slot,
>  	.digital_mute	= cpcap_voice_set_mute,
>  };
>  
> 

- Péter

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