Message ID | 20250404002728.3590501-1-quic_wcheng@quicinc.com |
---|---|
Headers | show |
Series | Introduce QC USB SND audio offloading support | expand |
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@quicinc.com> > --- > 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
Hi Stephan, On 4/4/2025 2:24 AM, Stephan Gerhold wrote: > 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@quicinc.com> >> --- >> 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 > Done >> + 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". > Sorry, missed that part. Fixed the naming and position of this. Thanks Wesley Cheng