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
