| Message ID | 20250319005141.312805-23-quic_wcheng@quicinc.com (mailing list archive) |
|---|---|
| State | New |
| Headers | show |
| Series | Introduce QC USB SND audio offloading support | expand |
On Tue, Mar 18, 2025 at 05:51:32PM -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 | 32 +++- > 6 files changed, 341 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 > [...] > @@ -513,12 +520,96 @@ struct afe_param_id_cdc_dma_cfg { > u16 active_channels_mask; > } __packed; > > +struct afe_param_id_usb_cfg { > +/* Minor version used for tracking USB audio device configuration. > + * Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG > + */ > + u32 cfg_minor_version; > +/* Sampling rate of the port. > + * Supported values: > + * - AFE_PORT_SAMPLE_RATE_8K > + * - AFE_PORT_SAMPLE_RATE_11025 > + * - AFE_PORT_SAMPLE_RATE_12K > + * - AFE_PORT_SAMPLE_RATE_16K > + * - AFE_PORT_SAMPLE_RATE_22050 > + * - AFE_PORT_SAMPLE_RATE_24K > + * - AFE_PORT_SAMPLE_RATE_32K > + * - AFE_PORT_SAMPLE_RATE_44P1K > + * - AFE_PORT_SAMPLE_RATE_48K > + * - AFE_PORT_SAMPLE_RATE_96K > + * - AFE_PORT_SAMPLE_RATE_192K > + */ > + u32 sample_rate; > +/* Bit width of the sample. > + * Supported values: 16, 24 > + */ > + u16 bit_width; > +/* Number of channels. > + * Supported values: 1 and 2 > + */ > + u16 num_channels; > +/* Data format supported by the USB. The supported value is > + * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM). > + */ > + u16 data_format; > +/* this field must be 0 */ > + u16 reserved; > +/* device token of actual end USB audio device */ > + u32 dev_token; > +/* endianness of this interface */ > + u32 endian; Nitpick: The indentation between u32 and the struct field names is odd, can you use a single tab character like in the afe_param_id_cdc_dma_cfg instead? > +/* service interval */ > + u32 service_interval; > +} __packed; > + > + [...] > diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c > index 4919001de08b..4a96b11f7fd1 100644 > --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c > +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c > @@ -97,6 +97,26 @@ > } > > static struct snd_soc_dai_driver q6dsp_audio_fe_dais[] = { > + { > + .playback = { > + .stream_name = "USB Playback", > + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | > + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | > + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | > + SNDRV_PCM_RATE_192000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | > + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | > + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | > + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE, > + .channels_min = 1, > + .channels_max = 2, > + .rate_min = 8000, > + .rate_max = 192000, Nitpick: Indentation after rate_max is also odd here, please choose one of the styles, either .rate_min = 8000, or .rate_max = 192000, > + }, > + .id = USB_RX, > + .name = "USB_RX", > + }, > { > .playback = { > .stream_name = "HDMI Playback", > [...] > diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c > index 90228699ba7d..b7439420b425 100644 > --- a/sound/soc/qcom/qdsp6/q6routing.c > +++ b/sound/soc/qcom/qdsp6/q6routing.c > @@ -435,6 +435,26 @@ static struct session_data *get_session_from_id(struct msm_routing_data *data, > > return NULL; > } > + > +static bool is_usb_routing_enabled(struct msm_routing_data *data) > +{ > + int i; > + > + /* > + * Loop through current sessions to see if there are active routes > + * to the USB_RX backend DAI. The USB offload routing is designed > + * similarly to the non offload path. If there are multiple PCM > + * devices associated with the ASoC platform card, only one active > + * path can be routed to the USB offloaded endpoint. > + */ > + for (i = 0; i < MAX_SESSIONS; i++) { > + if (data->sessions[i].port_id == USB_RX) > + return true; > + } > + > + return false; > +} What is different about USB_RX compared to other output ports we have in Q6AFE? Obviously, we can only play one stream on an output port. But doesn't the ADSP mix streams together when you have multiple routes? Also, this doesn't actually check for *active* routes only. It just looks if any other MultiMedia DAI is configured to output to USB_RX. That doesn't mean they will ever be active at the same time. I might for example want to have MultiMedia1 and MultiMedia2 both configured to output to USB_RX. Let's assume MultiMedia1 is a normal PCM DAI, MultiMedia2 is a compress offload DAI. When I want to playback normal audio, I go through MultiMedia1, when I want to play compressed audio, I go through MultiMedia2. Only one of them active at a time. Why can't I set this up statically in the mixers? If you confirm that it is really impossible to have multiple streams mixed together to the USB_RX output in the ADSP, then this should be a runtime check instead when starting the stream IMO. > + > /** > * q6routing_stream_close() - Deregister a stream > * > @@ -499,7 +519,8 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, > struct session_data *session = &data->sessions[session_id]; > > if (ucontrol->value.integer.value[0]) { > - if (session->port_id == be_id) > + if (session->port_id == be_id || > + (be_id == USB_RX && is_usb_routing_enabled(data))) > return 0; > > session->port_id = be_id; > @@ -515,6 +536,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, > return 1; > } > > +static const struct snd_kcontrol_new usb_mixer_controls[] = { usb_rx_mixer_controls > + Q6ROUTING_RX_MIXERS(USB_RX) }; > + > static const struct snd_kcontrol_new hdmi_mixer_controls[] = { > Q6ROUTING_RX_MIXERS(HDMI_RX) }; > > @@ -950,6 +974,10 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { > SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0, > mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)), > > + SND_SOC_DAPM_MIXER("USB Mixer", SND_SOC_NOPM, 0, 0, > + usb_mixer_controls, > + ARRAY_SIZE(usb_mixer_controls)), Please put this next to the other playback mixers above (below "RX_CODEC_DMA_RX_7 Audio Mixer"). I think it would also be more clear if you call this "USB_RX 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. Are you planning to send follow-up patches for USB recording offload (USB_TX) later? Me and Luca successfully used your series to playback voice call audio via the ADSP to an USB headset, recording would be also needed to use this fully. :-) Thanks, Stephan
Hi Stephan, On 3/25/2025 2:24 AM, Stephan Gerhold wrote: > On Tue, Mar 18, 2025 at 05:51:32PM -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 | 32 +++- >> 6 files changed, 341 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 >> [...] >> @@ -513,12 +520,96 @@ struct afe_param_id_cdc_dma_cfg { >> u16 active_channels_mask; >> } __packed; >> >> +struct afe_param_id_usb_cfg { >> +/* Minor version used for tracking USB audio device configuration. >> + * Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG >> + */ >> + u32 cfg_minor_version; >> +/* Sampling rate of the port. >> + * Supported values: >> + * - AFE_PORT_SAMPLE_RATE_8K >> + * - AFE_PORT_SAMPLE_RATE_11025 >> + * - AFE_PORT_SAMPLE_RATE_12K >> + * - AFE_PORT_SAMPLE_RATE_16K >> + * - AFE_PORT_SAMPLE_RATE_22050 >> + * - AFE_PORT_SAMPLE_RATE_24K >> + * - AFE_PORT_SAMPLE_RATE_32K >> + * - AFE_PORT_SAMPLE_RATE_44P1K >> + * - AFE_PORT_SAMPLE_RATE_48K >> + * - AFE_PORT_SAMPLE_RATE_96K >> + * - AFE_PORT_SAMPLE_RATE_192K >> + */ >> + u32 sample_rate; >> +/* Bit width of the sample. >> + * Supported values: 16, 24 >> + */ >> + u16 bit_width; >> +/* Number of channels. >> + * Supported values: 1 and 2 >> + */ >> + u16 num_channels; >> +/* Data format supported by the USB. The supported value is >> + * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM). >> + */ >> + u16 data_format; >> +/* this field must be 0 */ >> + u16 reserved; >> +/* device token of actual end USB audio device */ >> + u32 dev_token; >> +/* endianness of this interface */ >> + u32 endian; > > Nitpick: The indentation between u32 and the struct field names is odd, > can you use a single tab character like in the afe_param_id_cdc_dma_cfg > instead? > Ack. >> +/* service interval */ >> + u32 service_interval; >> +} __packed; >> + >> + [...] >> diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c >> index 4919001de08b..4a96b11f7fd1 100644 >> --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c >> +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c >> @@ -97,6 +97,26 @@ >> } >> >> static struct snd_soc_dai_driver q6dsp_audio_fe_dais[] = { >> + { >> + .playback = { >> + .stream_name = "USB Playback", >> + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | >> + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | >> + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | >> + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | >> + SNDRV_PCM_RATE_192000, >> + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | >> + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | >> + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | >> + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE, >> + .channels_min = 1, >> + .channels_max = 2, >> + .rate_min = 8000, >> + .rate_max = 192000, > > Nitpick: Indentation after rate_max is also odd here, please choose one > of the styles, either > > .rate_min = 8000, > > or > > .rate_max = 192000, > Ack. >> + }, >> + .id = USB_RX, >> + .name = "USB_RX", >> + }, >> { >> .playback = { >> .stream_name = "HDMI Playback", >> [...] >> diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c >> index 90228699ba7d..b7439420b425 100644 >> --- a/sound/soc/qcom/qdsp6/q6routing.c >> +++ b/sound/soc/qcom/qdsp6/q6routing.c >> @@ -435,6 +435,26 @@ static struct session_data *get_session_from_id(struct msm_routing_data *data, >> >> return NULL; >> } >> + >> +static bool is_usb_routing_enabled(struct msm_routing_data *data) >> +{ >> + int i; >> + >> + /* >> + * Loop through current sessions to see if there are active routes >> + * to the USB_RX backend DAI. The USB offload routing is designed >> + * similarly to the non offload path. If there are multiple PCM >> + * devices associated with the ASoC platform card, only one active >> + * path can be routed to the USB offloaded endpoint. >> + */ >> + for (i = 0; i < MAX_SESSIONS; i++) { >> + if (data->sessions[i].port_id == USB_RX) >> + return true; >> + } >> + >> + return false; >> +} > > What is different about USB_RX compared to other output ports we have in > Q6AFE? Obviously, we can only play one stream on an output port. But > doesn't the ADSP mix streams together when you have multiple routes? > This patch will limit the USB_RX from being able to be mixed to multiple q6adm paths. > Also, this doesn't actually check for *active* routes only. It just > looks if any other MultiMedia DAI is configured to output to USB_RX. > That doesn't mean they will ever be active at the same time. > Yes, the main reason being that that is the mechanism we use to populate the active offload path within the USB SND card mixer. > I might for example want to have MultiMedia1 and MultiMedia2 both > configured to output to USB_RX. Let's assume MultiMedia1 is a normal PCM > DAI, MultiMedia2 is a compress offload DAI. When I want to playback > normal audio, I go through MultiMedia1, when I want to play compressed > audio, I go through MultiMedia2. Only one of them active at a time. > Why can't I set this up statically in the mixers? > > If you confirm that it is really impossible to have multiple streams > mixed together to the USB_RX output in the ADSP, then this should be a > runtime check instead when starting the stream IMO. > We can have multiple streams being mixed together, but it will get confusing because it changes the definition that we had discussed about in the past about the overall design for the interaction w/ userspace. Although we (QC) only support a single USB audio device for offloading, there could be other situations where the audio DSP can support multiple devices. The assumption is that each MM path is assigned to a USB device. >> + >> /** >> * q6routing_stream_close() - Deregister a stream >> * >> @@ -499,7 +519,8 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, >> struct session_data *session = &data->sessions[session_id]; >> >> if (ucontrol->value.integer.value[0]) { >> - if (session->port_id == be_id) >> + if (session->port_id == be_id || >> + (be_id == USB_RX && is_usb_routing_enabled(data))) >> return 0; >> >> session->port_id = be_id; >> @@ -515,6 +536,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, >> return 1; >> } >> >> +static const struct snd_kcontrol_new usb_mixer_controls[] = { > > usb_rx_mixer_controls > >> + Q6ROUTING_RX_MIXERS(USB_RX) }; >> + >> static const struct snd_kcontrol_new hdmi_mixer_controls[] = { >> Q6ROUTING_RX_MIXERS(HDMI_RX) }; >> >> @@ -950,6 +974,10 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { >> SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0, >> mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)), >> >> + SND_SOC_DAPM_MIXER("USB Mixer", SND_SOC_NOPM, 0, 0, >> + usb_mixer_controls, >> + ARRAY_SIZE(usb_mixer_controls)), > > Please put this next to the other playback mixers above (below > "RX_CODEC_DMA_RX_7 Audio Mixer"). > > I think it would also be more clear if you call this "USB_RX 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. > Sure, will do. > > Are you planning to send follow-up patches for USB recording offload > (USB_TX) later? Me and Luca successfully used your series to playback > voice call audio via the ADSP to an USB headset, recording would be also > needed to use this fully. :-) > Yes, I will follow up after getting the bulk of the changes for playback merged first. The TX side changes should be minimal, and require only small updates. Thanks Wesley Cheng
On Tue, Mar 25, 2025 at 04:18:03PM -0700, Wesley Cheng wrote: > On 3/25/2025 2:24 AM, Stephan Gerhold wrote: > > On Tue, Mar 18, 2025 at 05:51:32PM -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 | 32 +++- > >> 6 files changed, 341 insertions(+), 3 deletions(-) > >> > [...] > >> diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c > >> index 90228699ba7d..b7439420b425 100644 > >> --- a/sound/soc/qcom/qdsp6/q6routing.c > >> +++ b/sound/soc/qcom/qdsp6/q6routing.c > >> @@ -435,6 +435,26 @@ static struct session_data *get_session_from_id(struct msm_routing_data *data, > >> > >> return NULL; > >> } > >> + > >> +static bool is_usb_routing_enabled(struct msm_routing_data *data) > >> +{ > >> + int i; > >> + > >> + /* > >> + * Loop through current sessions to see if there are active routes > >> + * to the USB_RX backend DAI. The USB offload routing is designed > >> + * similarly to the non offload path. If there are multiple PCM > >> + * devices associated with the ASoC platform card, only one active > >> + * path can be routed to the USB offloaded endpoint. > >> + */ > >> + for (i = 0; i < MAX_SESSIONS; i++) { > >> + if (data->sessions[i].port_id == USB_RX) > >> + return true; > >> + } > >> + > >> + return false; > >> +} > > > > What is different about USB_RX compared to other output ports we have in > > Q6AFE? Obviously, we can only play one stream on an output port. But > > doesn't the ADSP mix streams together when you have multiple routes? > > > > This patch will limit the USB_RX from being able to be mixed to multiple > q6adm paths. > > > Also, this doesn't actually check for *active* routes only. It just > > looks if any other MultiMedia DAI is configured to output to USB_RX. > > That doesn't mean they will ever be active at the same time. > > > > Yes, the main reason being that that is the mechanism we use to populate > the active offload path within the USB SND card mixer. > > > I might for example want to have MultiMedia1 and MultiMedia2 both > > configured to output to USB_RX. Let's assume MultiMedia1 is a normal PCM > > DAI, MultiMedia2 is a compress offload DAI. When I want to playback > > normal audio, I go through MultiMedia1, when I want to play compressed > > audio, I go through MultiMedia2. Only one of them active at a time. > > Why can't I set this up statically in the mixers? > > > > If you confirm that it is really impossible to have multiple streams > > mixed together to the USB_RX output in the ADSP, then this should be a > > runtime check instead when starting the stream IMO. > > > > We can have multiple streams being mixed together, but it will get > confusing because it changes the definition that we had discussed about in > the past about the overall design for the interaction w/ userspace. > Although we (QC) only support a single USB audio device for offloading, > there could be other situations where the audio DSP can support multiple > devices. The assumption is that each MM path is assigned to a USB device. > Are you referring to the "USB Offload Playback Route PCM#*" mixers here? They could just refer to first of the configured MM paths, if someone decides to route multiple paths to the USB backend. Looking at q6usb_update_offload_route(), I think the implementation does that already. I think it's fine that the userspace API for automatically "probing" the PCM device supports only a single path to the USB backend. But if someone wants to bypass the automatic probing and configure a more advanced setup, do we need to forbid that? Asked differently: what would happen if we remove this check here and handle USB_RX like any other Q6AFE output port? Would anything break for the userspace interface? > [...] > > > > Are you planning to send follow-up patches for USB recording offload > > (USB_TX) later? Me and Luca successfully used your series to playback > > voice call audio via the ADSP to an USB headset, recording would be also > > needed to use this fully. :-) > > > > Yes, I will follow up after getting the bulk of the changes for playback > merged first. The TX side changes should be minimal, and require only > small updates. > Thanks, sounds good! Stephan
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 @@ -92,6 +92,39 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } +static int q6afe_usb_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + int channels = params_channels(params); + int rate = params_rate(params); + struct q6afe_usb_cfg *usb = &dai_data->port_config[dai->id].usb_audio; + + usb->sample_rate = rate; + usb->num_channels = channels; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + usb->bit_width = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: + usb->bit_width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + usb->bit_width = 32; + break; + default: + dev_err(dai->dev, "%s: invalid format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + return 0; +} + static int q6i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -394,6 +427,10 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream, q6afe_cdc_dma_port_prepare(dai_data->port[dai->id], &dai_data->port_config[dai->id].dma_cfg); break; + case USB_RX: + q6afe_usb_port_prepare(dai_data->port[dai->id], + &dai_data->port_config[dai->id].usb_audio); + break; default: return -EINVAL; } @@ -622,6 +659,9 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"TX_CODEC_DMA_TX_5", NULL, "TX_CODEC_DMA_TX_5 Capture"}, {"RX_CODEC_DMA_RX_6 Playback", NULL, "RX_CODEC_DMA_RX_6"}, {"RX_CODEC_DMA_RX_7 Playback", NULL, "RX_CODEC_DMA_RX_7"}, + + /* USB playback AFE port receives data for playback, hence use the RX port */ + {"USB Playback", NULL, "USB_RX"}, }; static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai) @@ -649,6 +689,23 @@ static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai) return 0; } +static const struct snd_soc_dai_ops q6afe_usb_ops = { + .probe = msm_dai_q6_dai_probe, + .prepare = q6afe_dai_prepare, + .hw_params = q6afe_usb_hw_params, + /* + * Shutdown callback required to stop the USB AFE port, which is enabled + * by the prepare() stage. This stops the audio traffic on the USB AFE + * port on the Q6DSP. + */ + .shutdown = q6afe_dai_shutdown, + /* + * Startup callback not needed, as AFE port start command passes the PCM + * parameters within the AFE command, which is provided by the PCM core + * during the prepare() stage. + */ +}; + static const struct snd_soc_dai_ops q6hdmi_ops = { .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, @@ -947,6 +1004,8 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_7", "NULL", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_IN("USB_RX", NULL, 0, SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_component_driver q6afe_dai_component = { @@ -1061,6 +1120,7 @@ static int q6afe_dai_dev_probe(struct platform_device *pdev) cfg.q6i2s_ops = &q6i2s_ops; cfg.q6tdm_ops = &q6tdm_ops; cfg.q6dma_ops = &q6dma_ops; + cfg.q6usb_ops = &q6afe_usb_ops; dais = q6dsp_audio_ports_set_config(dev, &cfg, &num_dais); return devm_snd_soc_register_component(dev, &q6afe_dai_component, dais, num_dais); diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index ef7557be5d66..3ee6ff8a6bc3 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -35,6 +35,8 @@ #define AFE_MODULE_TDM 0x0001028A #define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235 +#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS 0x000102A5 +#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA #define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238 #define AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG 0x00010239 @@ -44,6 +46,7 @@ #define AFE_PARAM_ID_TDM_CONFIG 0x0001029D #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297 #define AFE_PARAM_ID_CODEC_DMA_CONFIG 0x000102B8 +#define AFE_PARAM_ID_USB_AUDIO_CONFIG 0x000102A4 #define AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST 0x000100f4 #define AFE_CMD_RSP_REMOTE_LPASS_CORE_HW_VOTE_REQUEST 0x000100f5 #define AFE_CMD_REMOTE_LPASS_CORE_HW_DEVOTE_REQUEST 0x000100f6 @@ -72,12 +75,16 @@ #define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1 #define AFE_LINEAR_PCM_DATA 0x0 +#define AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG 0x1 /* Port IDs */ #define AFE_API_VERSION_HDMI_CONFIG 0x1 #define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E #define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020 +/* USB AFE port */ +#define AFE_PORT_ID_USB_RX 0x7000 + #define AFE_API_VERSION_SLIMBUS_CONFIG 0x1 /* Clock set API version */ #define AFE_API_VERSION_CLOCK_SET 1 @@ -513,12 +520,96 @@ struct afe_param_id_cdc_dma_cfg { u16 active_channels_mask; } __packed; +struct afe_param_id_usb_cfg { +/* Minor version used for tracking USB audio device configuration. + * Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + */ + u32 cfg_minor_version; +/* Sampling rate of the port. + * Supported values: + * - AFE_PORT_SAMPLE_RATE_8K + * - AFE_PORT_SAMPLE_RATE_11025 + * - AFE_PORT_SAMPLE_RATE_12K + * - AFE_PORT_SAMPLE_RATE_16K + * - AFE_PORT_SAMPLE_RATE_22050 + * - AFE_PORT_SAMPLE_RATE_24K + * - AFE_PORT_SAMPLE_RATE_32K + * - AFE_PORT_SAMPLE_RATE_44P1K + * - AFE_PORT_SAMPLE_RATE_48K + * - AFE_PORT_SAMPLE_RATE_96K + * - AFE_PORT_SAMPLE_RATE_192K + */ + u32 sample_rate; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + u16 bit_width; +/* Number of channels. + * Supported values: 1 and 2 + */ + u16 num_channels; +/* Data format supported by the USB. The supported value is + * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM). + */ + u16 data_format; +/* this field must be 0 */ + u16 reserved; +/* device token of actual end USB audio device */ + u32 dev_token; +/* endianness of this interface */ + u32 endian; +/* service interval */ + u32 service_interval; +} __packed; + +/** + * struct afe_param_id_usb_audio_dev_params + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @dev_token: device token of actual end USB audio device + **/ +struct afe_param_id_usb_audio_dev_params { + u32 cfg_minor_version; + u32 dev_token; +} __packed; + +/** + * struct afe_param_id_usb_audio_dev_lpcm_fmt + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @endian: endianness of this interface + **/ +struct afe_param_id_usb_audio_dev_lpcm_fmt { + u32 cfg_minor_version; + u32 endian; +} __packed; + +#define AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL 0x000102B7 + +/** + * struct afe_param_id_usb_audio_svc_interval + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @svc_interval: service interval + **/ +struct afe_param_id_usb_audio_svc_interval { + u32 cfg_minor_version; + u32 svc_interval; +} __packed; + union afe_port_config { struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch; struct afe_param_id_slimbus_cfg slim_cfg; struct afe_param_id_i2s_cfg i2s_cfg; struct afe_param_id_tdm_cfg tdm_cfg; struct afe_param_id_cdc_dma_cfg dma_cfg; + struct afe_param_id_usb_cfg usb_cfg; } __packed; @@ -833,6 +924,7 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { RX_CODEC_DMA_RX_6, 1, 1}, [RX_CODEC_DMA_RX_7] = { AFE_PORT_ID_RX_CODEC_DMA_RX_7, RX_CODEC_DMA_RX_7, 1, 1}, + [USB_RX] = { AFE_PORT_ID_USB_RX, USB_RX, 1, 1}, }; static void q6afe_port_free(struct kref *ref) @@ -1290,6 +1382,101 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port, } EXPORT_SYMBOL_GPL(q6afe_tdm_port_prepare); +/** + * afe_port_send_usb_dev_param() - Send USB dev token + * + * @port: Instance of afe port + * @cardidx: USB SND card index to reference + * @pcmidx: USB SND PCM device index to reference + * + * The USB dev token carries information about which USB SND card instance and + * PCM device to execute the offload on. This information is carried through + * to the stream enable QMI request, which is handled by the offload class + * driver. The information is parsed to determine which USB device to query + * the required resources for. + */ +int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx) +{ + struct afe_param_id_usb_audio_dev_params usb_dev; + int ret; + + memset(&usb_dev, 0, sizeof(usb_dev)); + + usb_dev.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; + usb_dev.dev_token = (cardidx << 16) | (pcmidx << 8); + ret = q6afe_port_set_param_v2(port, &usb_dev, + AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS, + AFE_MODULE_AUDIO_DEV_INTERFACE, + sizeof(usb_dev)); + if (ret) + dev_err(port->afe->dev, "%s: AFE device param cmd failed %d\n", + __func__, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(afe_port_send_usb_dev_param); + +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; + } + + 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; + } + + 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: + return ret; +} + +/** + * q6afe_usb_port_prepare() - Prepare usb afe port. + * + * @port: Instance of afe port + * @cfg: USB configuration for the afe port + * + */ +void q6afe_usb_port_prepare(struct q6afe_port *port, + struct q6afe_usb_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + + pcfg->usb_cfg.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; + pcfg->usb_cfg.sample_rate = cfg->sample_rate; + pcfg->usb_cfg.num_channels = cfg->num_channels; + pcfg->usb_cfg.bit_width = cfg->bit_width; + + afe_port_send_usb_params(port, cfg); +} +EXPORT_SYMBOL_GPL(q6afe_usb_port_prepare); + /** * q6afe_hdmi_port_prepare() - Prepare hdmi afe port. * @@ -1612,7 +1799,10 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) break; case AFE_PORT_ID_WSA_CODEC_DMA_RX_0 ... AFE_PORT_ID_RX_CODEC_DMA_RX_7: cfg_type = AFE_PARAM_ID_CODEC_DMA_CONFIG; - break; + break; + case AFE_PORT_ID_USB_RX: + cfg_type = AFE_PARAM_ID_USB_AUDIO_CONFIG; + break; default: dev_err(dev, "Invalid port id 0x%x\n", port_id); return ERR_PTR(-EINVAL); diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 65d0676075e1..a29abe4ce436 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -3,7 +3,7 @@ #ifndef __Q6AFE_H__ #define __Q6AFE_H__ -#define AFE_PORT_MAX 129 +#define AFE_PORT_MAX 137 #define MSM_AFE_PORT_TYPE_RX 0 #define MSM_AFE_PORT_TYPE_TX 1 @@ -203,6 +203,36 @@ struct q6afe_cdc_dma_cfg { u16 active_channels_mask; }; +/** + * struct q6afe_usb_cfg + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @sample_rate: Sampling rate of the port + * Supported values: + * AFE_PORT_SAMPLE_RATE_8K + * AFE_PORT_SAMPLE_RATE_11025 + * AFE_PORT_SAMPLE_RATE_12K + * AFE_PORT_SAMPLE_RATE_16K + * AFE_PORT_SAMPLE_RATE_22050 + * AFE_PORT_SAMPLE_RATE_24K + * AFE_PORT_SAMPLE_RATE_32K + * AFE_PORT_SAMPLE_RATE_44P1K + * AFE_PORT_SAMPLE_RATE_48K + * AFE_PORT_SAMPLE_RATE_96K + * AFE_PORT_SAMPLE_RATE_192K + * @bit_width: Bit width of the sample. + * Supported values: 16, 24 + * @num_channels: Number of channels + * Supported values: 1, 2 + **/ +struct q6afe_usb_cfg { + u32 cfg_minor_version; + u32 sample_rate; + u16 bit_width; + u16 num_channels; +}; struct q6afe_port_config { struct q6afe_hdmi_cfg hdmi; @@ -210,6 +240,7 @@ struct q6afe_port_config { struct q6afe_i2s_cfg i2s_cfg; struct q6afe_tdm_cfg tdm; struct q6afe_cdc_dma_cfg dma_cfg; + struct q6afe_usb_cfg usb_audio; }; struct q6afe_port; @@ -219,6 +250,8 @@ int q6afe_port_start(struct q6afe_port *port); int q6afe_port_stop(struct q6afe_port *port); void q6afe_port_put(struct q6afe_port *port); int q6afe_get_port_id(int index); +void q6afe_usb_port_prepare(struct q6afe_port *port, + struct q6afe_usb_cfg *cfg); void q6afe_hdmi_port_prepare(struct q6afe_port *port, struct q6afe_hdmi_cfg *cfg); void q6afe_slim_port_prepare(struct q6afe_port *port, @@ -228,6 +261,7 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg); void q6afe_cdc_dma_port_prepare(struct q6afe_port *port, struct q6afe_cdc_dma_cfg *cfg); +int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx); int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id, int clk_src, int clk_root, unsigned int freq, int dir); diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c index 4919001de08b..4a96b11f7fd1 100644 --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c @@ -97,6 +97,26 @@ } static struct snd_soc_dai_driver q6dsp_audio_fe_dais[] = { + { + .playback = { + .stream_name = "USB Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE, + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 192000, + }, + .id = USB_RX, + .name = "USB_RX", + }, { .playback = { .stream_name = "HDMI Playback", @@ -624,6 +644,9 @@ struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev, case WSA_CODEC_DMA_RX_0 ... RX_CODEC_DMA_RX_7: q6dsp_audio_fe_dais[i].ops = cfg->q6dma_ops; break; + case USB_RX: + q6dsp_audio_fe_dais[i].ops = cfg->q6usb_ops; + break; default: break; } diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h index 7f052c8a1257..d8dde6dd0aca 100644 --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h @@ -11,6 +11,7 @@ struct q6dsp_audio_port_dai_driver_config { const struct snd_soc_dai_ops *q6i2s_ops; const struct snd_soc_dai_ops *q6tdm_ops; const struct snd_soc_dai_ops *q6dma_ops; + const struct snd_soc_dai_ops *q6usb_ops; }; struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev, diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 90228699ba7d..b7439420b425 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -435,6 +435,26 @@ static struct session_data *get_session_from_id(struct msm_routing_data *data, return NULL; } + +static bool is_usb_routing_enabled(struct msm_routing_data *data) +{ + int i; + + /* + * Loop through current sessions to see if there are active routes + * to the USB_RX backend DAI. The USB offload routing is designed + * similarly to the non offload path. If there are multiple PCM + * devices associated with the ASoC platform card, only one active + * path can be routed to the USB offloaded endpoint. + */ + for (i = 0; i < MAX_SESSIONS; i++) { + if (data->sessions[i].port_id == USB_RX) + return true; + } + + return false; +} + /** * q6routing_stream_close() - Deregister a stream * @@ -499,7 +519,8 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, struct session_data *session = &data->sessions[session_id]; if (ucontrol->value.integer.value[0]) { - if (session->port_id == be_id) + if (session->port_id == be_id || + (be_id == USB_RX && is_usb_routing_enabled(data))) return 0; session->port_id = be_id; @@ -515,6 +536,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, return 1; } +static const struct snd_kcontrol_new usb_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(USB_RX) }; + static const struct snd_kcontrol_new hdmi_mixer_controls[] = { Q6ROUTING_RX_MIXERS(HDMI_RX) }; @@ -950,6 +974,10 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0, mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)), + SND_SOC_DAPM_MIXER("USB Mixer", SND_SOC_NOPM, 0, 0, + usb_mixer_controls, + ARRAY_SIZE(usb_mixer_controls)), + }; static const struct snd_soc_dapm_route intercon[] = { @@ -1043,6 +1071,8 @@ 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"), }; static int routing_hw_params(struct snd_soc_component *component,
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 | 32 +++- 6 files changed, 341 insertions(+), 3 deletions(-)