Re: [PATCH v37 22/31] ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp

[Stephan Gerhold]

On Thu, Apr 03, 2025 at 05:27:19PM -0700, Wesley Cheng wrote:
> The QC ADSP is able to support USB playback endpoints, so that the main
> application processor can be placed into lower CPU power modes.  This adds
> the required AFE port configurations and port start command to start an
> audio session.
> 
> Specifically, the QC ADSP can support all potential endpoints that are
> exposed by the audio data interface.  This includes isochronous data
> endpoints, in either synchronous mode or asynchronous mode. In the latter
> case both implicit or explicit feedback endpoints are supported.  The size
> of audio samples sent per USB frame (microframe) will be adjusted based on
> information received on the feedback endpoint.
> 
> Some pre-requisites are needed before issuing the AFE port start command,
> such as setting the USB AFE dev_token.  This carries information about the
> available USB SND cards and PCM devices that have been discovered on the
> USB bus.  The dev_token field is used by the audio DSP to notify the USB
> offload driver of which card and PCM index to enable playback on.
> 
> Signed-off-by: Wesley Cheng <quic_wcheng@xxxxxxxxxxx>
> ---
>  sound/soc/qcom/qdsp6/q6afe-dai.c         |  60 +++++++
>  sound/soc/qcom/qdsp6/q6afe.c             | 192 ++++++++++++++++++++++-
>  sound/soc/qcom/qdsp6/q6afe.h             |  36 ++++-
>  sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c |  23 +++
>  sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h |   1 +
>  sound/soc/qcom/qdsp6/q6routing.c         |  10 +-
>  6 files changed, 319 insertions(+), 3 deletions(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
> index 7d9628cda875..0f47aadaabe1 100644
> --- a/sound/soc/qcom/qdsp6/q6afe-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
> [...]
> +static int afe_port_send_usb_params(struct q6afe_port *port, struct q6afe_usb_cfg *cfg)
> +{
> +	union afe_port_config *pcfg = &port->port_cfg;
> +	struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt;
> +	struct afe_param_id_usb_audio_svc_interval svc_int;
> +	int ret;
> +
> +	if (!pcfg) {
> +		dev_err(port->afe->dev, "%s: Error, no configuration data\n", __func__);
> +		ret = -EINVAL;
> +		goto exit;

Nitpick: drop the goto here, just do "return -EINVAL;"

> +	}
> +
> +	memset(&lpcm_fmt, 0, sizeof(lpcm_fmt));
> +	memset(&svc_int, 0, sizeof(svc_int));
> +
> +	lpcm_fmt.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
> +	lpcm_fmt.endian = pcfg->usb_cfg.endian;
> +	ret = q6afe_port_set_param_v2(port, &lpcm_fmt,
> +				      AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT,
> +				      AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(lpcm_fmt));
> +	if (ret) {
> +		dev_err(port->afe->dev, "%s: AFE device param cmd LPCM_FMT failed %d\n",
> +			__func__, ret);
> +		goto exit;

return ret;

> +	}
> +
> +	svc_int.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
> +	svc_int.svc_interval = pcfg->usb_cfg.service_interval;
> +	ret = q6afe_port_set_param_v2(port, &svc_int,
> +				      AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL,
> +				      AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(svc_int));
> +	if (ret)
> +		dev_err(port->afe->dev, "%s: AFE device param cmd svc_interval failed %d\n",
> +			__func__, ret);
> +
> +exit:

drop

> +	return ret;
> +}
> +
> [...]
> diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
> index 90228699ba7d..0def036ed3c9 100644
> --- a/sound/soc/qcom/qdsp6/q6routing.c
> +++ b/sound/soc/qcom/qdsp6/q6routing.c
> @@ -435,6 +435,7 @@ static struct session_data *get_session_from_id(struct msm_routing_data *data,
>  
>  	return NULL;
>  }
> +
>  /**
>   * q6routing_stream_close() - Deregister a stream
>   *
> @@ -515,6 +516,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
>  	return 1;
>  }
>  
> +static const struct snd_kcontrol_new usb_rx_mixer_controls[] = {
> +	Q6ROUTING_RX_MIXERS(USB_RX) };
> +
>  static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
>  	Q6ROUTING_RX_MIXERS(HDMI_RX) };
>  
> @@ -933,6 +937,9 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
>  	SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_7 Audio Mixer", SND_SOC_NOPM, 0, 0,
>  		rx_codec_dma_rx_7_mixer_controls,
>  		ARRAY_SIZE(rx_codec_dma_rx_7_mixer_controls)),
> +	SND_SOC_DAPM_MIXER("USB Mixer", SND_SOC_NOPM, 0, 0,

As I wrote on v36:

I think it would be more clear if you call this "USB_RX Audio Mixer"
instead for consistency with the other playback mixers. This would also
avoid confusion later when USB_TX is added in addition to USB_RX.

The "Audio" part in the name is redundant, but looks like all the other
playback mixers have it as well ...


> +			   usb_rx_mixer_controls,
> +			   ARRAY_SIZE(usb_rx_mixer_controls)),
>  	SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0,
>  		mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
>  	SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
> @@ -949,7 +956,6 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
>  		mmul7_mixer_controls, ARRAY_SIZE(mmul7_mixer_controls)),
>  	SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,
>  		mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),
> -
>  };
>  
>  static const struct snd_soc_dapm_route intercon[] = {
> @@ -1043,6 +1049,8 @@ static const struct snd_soc_dapm_route intercon[] = {
>  	{"MM_UL6", NULL, "MultiMedia6 Mixer"},
>  	{"MM_UL7", NULL, "MultiMedia7 Mixer"},
>  	{"MM_UL8", NULL, "MultiMedia8 Mixer"},
> +
> +	Q6ROUTING_RX_DAPM_ROUTE("USB Mixer", "USB_RX"),

Put this below "Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_7 Audio Mixer".

Thanks,
Stephan