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Message-ID: <ae2b7d9e-d05e-54ac-4f18-27cc8c4e81a0@ti.com>
Date: Wed, 12 Feb 2020 11:17:32 +0200
From: Peter Ujfalusi <peter.ujfalusi@...com>
To: Tony Lindgren <tony@...mide.com>, Mark Brown <broonie@...nel.org>
CC: Liam Girdwood <lgirdwood@...il.com>,
Jaroslav Kysela <perex@...ex.cz>,
Takashi Iwai <tiwai@...e.com>, <alsa-devel@...a-project.org>,
<linux-kernel@...r.kernel.org>, <linux-omap@...r.kernel.org>,
"Arthur D ." <spinal.by@...il.com>,
Merlijn Wajer <merlijn@...zup.org>,
Pavel Machek <pavel@....cz>, Sebastian Reichel <sre@...nel.org>
Subject: Re: [PATCH] ASoC: cpcap: Implement set_tdm_slot for voice call
support
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@...il.com>
> Cc: Merlijn Wajer <merlijn@...zup.org>
> Cc: Pavel Machek <pavel@....cz>
> Cc: Sebastian Reichel <sre@...nel.org>
> Signed-off-by: Tony Lindgren <tony@...mide.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
Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki.
Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki
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