| Message ID | 20221223233200.26089-3-quic_wcheng@quicinc.com (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | Introduce QC USB SND audio offloading support | expand |
On 12/23/22 17:31, Wesley Cheng wrote: > The QC ADSP is able to support USB playback and capture, 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. It would be good to clarify what sort of endpoints can be supported. I presume the SOF-synchronous case is handled, but how would you deal with async endpoints with feedback (be it explicit or implicit)? Note that it's very hard to make the decision not to support async endpoints, there are quite a few devices that are exposed as async to work around an obscure legacy issue on Windows but are really sof-synchronous endpoints that never ask for any change of pace. > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > .../sound/qcom,q6dsp-lpass-ports.h | 1 + > sound/soc/qcom/qdsp6/q6afe-dai.c | 47 +++++ > sound/soc/qcom/qdsp6/q6afe.c | 183 ++++++++++++++++++ > sound/soc/qcom/qdsp6/q6afe.h | 46 ++++- > sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 +++ > sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + > sound/soc/qcom/qdsp6/q6routing.c | 8 + > 7 files changed, 308 insertions(+), 1 deletion(-) > > diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h > index 9f7c5103bc82..746bc462bb2e 100644 > --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h > +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h > @@ -131,6 +131,7 @@ > #define RX_CODEC_DMA_RX_7 126 > #define QUINARY_MI2S_RX 127 > #define QUINARY_MI2S_TX 128 > +#define USB_RX 129 the commit message says the DSP can support Playback and capture, but here there's capture only ... > > static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { > + {"USB Playback", NULL, "USB_RX"}, ... but here RX means playback? I am not sure I get the convention on directions and what is actually supported? > +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 that aligns with my feedback on the cover letter, if you connect a device that can support from than 2 channels should the DSP even expose this DSP-optimized path? Oh and I forgot, what happens if there are multiple audio devices connected, can the DSP deal with all of them? If not, how is this handled? > + */ > + 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 aduio device */ typo: audio > + u32 dev_token; > +/* endianness of this interface */ > + u32 endian; Is this a USB concept? I can't recall having seen any parts of the USB audio class spec that the data can be big or little endian? > +/* service interval */ > + u32 service_interval; > +} __packed; > +int afe_port_send_usb_dev_param(struct q6afe_port *port, struct q6afe_usb_cfg *cfg) > +{ > + union afe_port_config *pcfg = &port->port_cfg; > + struct afe_param_id_usb_audio_dev_params usb_dev; > + struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt; > + struct afe_param_id_usb_audio_svc_interval svc_int; > + int ret = 0; > + > + if (!pcfg) { > + pr_err("%s: Error, no configuration data\n", __func__); can you use a dev_err() here and the rest of the code? > + ret = -EINVAL; > + goto exit; > + } > + > + memset(&usb_dev, 0, sizeof(usb_dev)); > + memset(&lpcm_fmt, 0, sizeof(lpcm_fmt)); > + > + usb_dev.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; > + 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) { > + pr_err("%s: AFE device param cmd failed %d\n", > + __func__, ret); > + goto exit; > + } > + > + 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) { > + pr_err("%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) { > + pr_err("%s: AFE device param cmd svc_interval failed %d\n", > + __func__, ret); > + ret = -EINVAL; > + goto exit; > + } > +exit: > + return ret; > +}
On 24/12/2022 00:31, Wesley Cheng wrote: > The QC ADSP is able to support USB playback and capture, 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. > > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > .../sound/qcom,q6dsp-lpass-ports.h | 1 + > sound/soc/qcom/qdsp6/q6afe-dai.c | 47 +++++ > sound/soc/qcom/qdsp6/q6afe.c | 183 ++++++++++++++++++ > sound/soc/qcom/qdsp6/q6afe.h | 46 ++++- > sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 +++ > sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + > sound/soc/qcom/qdsp6/q6routing.c | 8 + > 7 files changed, 308 insertions(+), 1 deletion(-) > > diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h > index 9f7c5103bc82..746bc462bb2e 100644 > --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h > +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h > @@ -131,6 +131,7 @@ > #define RX_CODEC_DMA_RX_7 126 > #define QUINARY_MI2S_RX 127 > #define QUINARY_MI2S_TX 128 > +#define USB_RX 129 > > #define LPASS_CLK_ID_PRI_MI2S_IBIT 1 Bindings are separate patches. Please split. Best regards, Krzysztof
Hi Pierre, On 1/4/2023 3:33 PM, Pierre-Louis Bossart wrote: > > > On 12/23/22 17:31, Wesley Cheng wrote: >> The QC ADSP is able to support USB playback and capture, 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. > > It would be good to clarify what sort of endpoints can be supported. I > presume the SOF-synchronous case is handled, but how would you deal with > async endpoints with feedback (be it explicit or implicit)? > Sure, both types of feedback endpoints are expected to be supported by the audio DSP, as well as sync eps. We have the logic there to modify the audio sample size accordingly. > Note that it's very hard to make the decision not to support async > endpoints, there are quite a few devices that are exposed as async to > work around an obscure legacy issue on Windows but are really > sof-synchronous endpoints that never ask for any change of pace. > >> Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> >> --- >> .../sound/qcom,q6dsp-lpass-ports.h | 1 + >> sound/soc/qcom/qdsp6/q6afe-dai.c | 47 +++++ >> sound/soc/qcom/qdsp6/q6afe.c | 183 ++++++++++++++++++ >> sound/soc/qcom/qdsp6/q6afe.h | 46 ++++- >> sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 +++ >> sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + >> sound/soc/qcom/qdsp6/q6routing.c | 8 + >> 7 files changed, 308 insertions(+), 1 deletion(-) >> >> diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h >> index 9f7c5103bc82..746bc462bb2e 100644 >> --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h >> +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h >> @@ -131,6 +131,7 @@ >> #define RX_CODEC_DMA_RX_7 126 >> #define QUINARY_MI2S_RX 127 >> #define QUINARY_MI2S_TX 128 >> +#define USB_RX 129 > > the commit message says the DSP can support Playback and capture, but > here there's capture only ... > > Sorry I will update the commit message properly. At the moment we've only verified audio playback. >> >> static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { >> + {"USB Playback", NULL, "USB_RX"}, > > ... but here RX means playback? > > I am not sure I get the convention on directions and what is actually > supported? > The notation is based on the direction of which the audio data is sourced or pushed on to the DSP. So in playback, the DSP is receiving audio data to send, and capture, it is transmitting audio data received. (although we do not support capture yet) >> +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 > > that aligns with my feedback on the cover letter, if you connect a > device that can support from than 2 channels should the DSP even expose > this DSP-optimized path? > My assumption is that I programmed the DAIs w/ PCM formats supported by the DSP, so I think the ASoC core should not allow userspace to choose that path if the hw params don't fit/match. > Oh and I forgot, what happens if there are multiple audio devices > connected, can the DSP deal with all of them? If not, how is this handled? > This is one topic that we were pretty open ended on. At least on our implementation, only one audio device can be supported at a time. We choose the latest device that was plugged in or discovered by the USB SND class driver. >> + */ >> + 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 aduio device */ > > typo: audio > Thanks >> + u32 dev_token; >> +/* endianness of this interface */ >> + u32 endian; > > Is this a USB concept? I can't recall having seen any parts of the USB > audio class spec that the data can be big or little endian? > No, this is probably just something our audio DSP uses on the AFE commands that it receives. >> +/* service interval */ >> + u32 service_interval; >> +} __packed; > >> +int afe_port_send_usb_dev_param(struct q6afe_port *port, struct q6afe_usb_cfg *cfg) >> +{ >> + union afe_port_config *pcfg = &port->port_cfg; >> + struct afe_param_id_usb_audio_dev_params usb_dev; >> + struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt; >> + struct afe_param_id_usb_audio_svc_interval svc_int; >> + int ret = 0; >> + >> + if (!pcfg) { >> + pr_err("%s: Error, no configuration data\n", __func__); > > can you use a dev_err() here and the rest of the code? > Sure. Thanks Wesley Cheng
Hi Krzysztof, On 1/5/2023 10:09 AM, Krzysztof Kozlowski wrote: > On 24/12/2022 00:31, Wesley Cheng wrote: >> The QC ADSP is able to support USB playback and capture, 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. >> >> Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> >> --- >> .../sound/qcom,q6dsp-lpass-ports.h | 1 + >> sound/soc/qcom/qdsp6/q6afe-dai.c | 47 +++++ >> sound/soc/qcom/qdsp6/q6afe.c | 183 ++++++++++++++++++ >> sound/soc/qcom/qdsp6/q6afe.h | 46 ++++- >> sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 +++ >> sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + >> sound/soc/qcom/qdsp6/q6routing.c | 8 + >> 7 files changed, 308 insertions(+), 1 deletion(-) >> >> diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h >> index 9f7c5103bc82..746bc462bb2e 100644 >> --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h >> +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h >> @@ -131,6 +131,7 @@ >> #define RX_CODEC_DMA_RX_7 126 >> #define QUINARY_MI2S_RX 127 >> #define QUINARY_MI2S_TX 128 >> +#define USB_RX 129 >> >> #define LPASS_CLK_ID_PRI_MI2S_IBIT 1 > > Bindings are separate patches. Please split. > Thanks for catching this. Will do. Thanks Wesley Cheng
>>> The QC ADSP is able to support USB playback and capture, 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. >> >> It would be good to clarify what sort of endpoints can be supported. I >> presume the SOF-synchronous case is handled, but how would you deal with >> async endpoints with feedback (be it explicit or implicit)? >> > > Sure, both types of feedback endpoints are expected to be supported by > the audio DSP, as well as sync eps. We have the logic there to modify > the audio sample size accordingly. did you mean modify samples per USB frame (or uframe), so as to change the pace at which data is transferred? If yes it'd be the same for Intel. >>> static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { >>> + {"USB Playback", NULL, "USB_RX"}, >> >> ... but here RX means playback? >> >> I am not sure I get the convention on directions and what is actually >> supported? >> > > The notation is based on the direction of which the audio data is > sourced or pushed on to the DSP. So in playback, the DSP is receiving > audio data to send, and capture, it is transmitting audio data received. > (although we do not support capture yet) ok, it'd be good to add a comment on this convention. Usually RX/TX is bus-centric. > >>> +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 >> >> that aligns with my feedback on the cover letter, if you connect a >> device that can support from than 2 channels should the DSP even expose >> this DSP-optimized path? >> > > My assumption is that I programmed the DAIs w/ PCM formats supported by > the DSP, so I think the ASoC core should not allow userspace to choose > that path if the hw params don't fit/match. Right, but the point I was trying to make is that if the device can do more, why create this DSP path at all? > >> Oh and I forgot, what happens if there are multiple audio devices >> connected, can the DSP deal with all of them? If not, how is this >> handled? >> > > This is one topic that we were pretty open ended on. At least on our > implementation, only one audio device can be supported at a time. We > choose the latest device that was plugged in or discovered by the USB > SND class driver. Similar case for Intel. I have to revisit this, I don't recall the details. >>> + u32 dev_token; >>> +/* endianness of this interface */ >>> + u32 endian; >> >> Is this a USB concept? I can't recall having seen any parts of the USB >> audio class spec that the data can be big or little endian? >> > > No, this is probably just something our audio DSP uses on the AFE > commands that it receives. ok.
Hi Pierre, On 1/6/2023 8:09 AM, Pierre-Louis Bossart wrote: > >>>> The QC ADSP is able to support USB playback and capture, 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. >>> >>> It would be good to clarify what sort of endpoints can be supported. I >>> presume the SOF-synchronous case is handled, but how would you deal with >>> async endpoints with feedback (be it explicit or implicit)? >>> >> >> Sure, both types of feedback endpoints are expected to be supported by >> the audio DSP, as well as sync eps. We have the logic there to modify >> the audio sample size accordingly. > > did you mean modify samples per USB frame (or uframe), so as to change > the pace at which data is transferred? If yes it'd be the same for Intel. > Yes, sorry for not being clear. Your understanding is correct. >>>> static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { >>>> + {"USB Playback", NULL, "USB_RX"}, >>> >>> ... but here RX means playback? >>> >>> I am not sure I get the convention on directions and what is actually >>> supported? >>> >> >> The notation is based on the direction of which the audio data is >> sourced or pushed on to the DSP. So in playback, the DSP is receiving >> audio data to send, and capture, it is transmitting audio data received. >> (although we do not support capture yet) > > ok, it'd be good to add a comment on this convention. Usually RX/TX is > bus-centric. > Sure, will do. >> >>>> +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 >>> >>> that aligns with my feedback on the cover letter, if you connect a >>> device that can support from than 2 channels should the DSP even expose >>> this DSP-optimized path? >>> >> >> My assumption is that I programmed the DAIs w/ PCM formats supported by >> the DSP, so I think the ASoC core should not allow userspace to choose >> that path if the hw params don't fit/match. > > Right, but the point I was trying to make is that if the device can do > more, why create this DSP path at all? > Yeah, I think this brings me back to needing to understand a bit more of how the userspace chooses which PCM device to use. At least for our current use cases, userspace would always route through the offload path, regardless of if the device can do more. It will just select a lower audio profile if so. >> >>> Oh and I forgot, what happens if there are multiple audio devices >>> connected, can the DSP deal with all of them? If not, how is this >>> handled? >>> >> >> This is one topic that we were pretty open ended on. At least on our >> implementation, only one audio device can be supported at a time. We >> choose the latest device that was plugged in or discovered by the USB >> SND class driver. > > Similar case for Intel. I have to revisit this, I don't recall the details. > Got it. Thanks Wesley Cheng
diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h index 9f7c5103bc82..746bc462bb2e 100644 --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h @@ -131,6 +131,7 @@ #define RX_CODEC_DMA_RX_7 126 #define QUINARY_MI2S_RX 127 #define QUINARY_MI2S_TX 128 +#define USB_RX 129 #define LPASS_CLK_ID_PRI_MI2S_IBIT 1 #define LPASS_CLK_ID_PRI_MI2S_EBIT 2 diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 8bb7452b8f18..7fdee8cbd4de 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -111,6 +111,40 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } +static int q6usb_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: + case SNDRV_PCM_FORMAT_SPECIAL: + 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) @@ -411,6 +445,10 @@ 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; } @@ -495,6 +533,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, } static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { + {"USB Playback", NULL, "USB_RX"}, {"HDMI Playback", NULL, "HDMI_RX"}, {"Display Port Playback", NULL, "DISPLAY_PORT_RX"}, {"Slimbus Playback", NULL, "SLIMBUS_0_RX"}, @@ -639,6 +678,12 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"RX_CODEC_DMA_RX_7 Playback", NULL, "RX_CODEC_DMA_RX_7"}, }; +static const struct snd_soc_dai_ops q6usb_ops = { + .prepare = q6afe_dai_prepare, + .hw_params = q6usb_hw_params, + .shutdown = q6afe_dai_shutdown, +}; + static const struct snd_soc_dai_ops q6hdmi_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6hdmi_hw_params, @@ -703,6 +748,7 @@ static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai) } static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { + SND_SOC_DAPM_AIF_IN("USB_RX", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, SND_SOC_NOPM, 0, 0), @@ -1068,6 +1114,7 @@ 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 = &q6usb_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 919e326b9462..2054f5723e03 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -34,6 +34,8 @@ #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 @@ -43,6 +45,7 @@ #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 @@ -71,12 +74,16 @@ #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 @@ -512,12 +519,109 @@ 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 aduio 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 aduio 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; + +/** + * struct afe_param_id_usb_audio_dev_latency_mode + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_LATENCY_MODE + * @mode: latency mode for the USB audio device + **/ +struct afe_param_id_usb_audio_dev_latency_mode { + u32 minor_version; + u32 mode; +} __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; @@ -577,6 +681,7 @@ struct afe_port_map { */ static struct afe_port_map port_maps[AFE_PORT_MAX] = { + [USB_RX] = { AFE_PORT_ID_USB_RX, USB_RX, 1, 1}, [HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1}, [SLIMBUS_0_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX, SLIMBUS_0_RX, 1, 1}, @@ -1289,6 +1394,82 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port, } EXPORT_SYMBOL_GPL(q6afe_tdm_port_prepare); +int afe_port_send_usb_dev_param(struct q6afe_port *port, struct q6afe_usb_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + struct afe_param_id_usb_audio_dev_params usb_dev; + struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt; + struct afe_param_id_usb_audio_svc_interval svc_int; + int ret = 0; + + if (!pcfg) { + pr_err("%s: Error, no configuration data\n", __func__); + ret = -EINVAL; + goto exit; + } + + memset(&usb_dev, 0, sizeof(usb_dev)); + memset(&lpcm_fmt, 0, sizeof(lpcm_fmt)); + + usb_dev.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; + 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) { + pr_err("%s: AFE device param cmd failed %d\n", + __func__, ret); + goto exit; + } + + 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) { + pr_err("%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) { + pr_err("%s: AFE device param cmd svc_interval failed %d\n", + __func__, ret); + ret = -EINVAL; + goto exit; + } +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_dev_param(port, cfg); +} +EXPORT_SYMBOL_GPL(q6afe_usb_port_prepare); + /** * q6afe_hdmi_port_prepare() - Prepare hdmi afe port. * @@ -1611,6 +1792,8 @@ 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; + 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); diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 30fd77e2f458..88550a08e57d 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -5,7 +5,7 @@ #include <dt-bindings/sound/qcom,q6afe.h> -#define AFE_PORT_MAX 129 +#define AFE_PORT_MAX 130 #define MSM_AFE_PORT_TYPE_RX 0 #define MSM_AFE_PORT_TYPE_TX 1 @@ -205,6 +205,47 @@ 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 + * @data_format: Data format supported by the USB + * Supported values: 0 + * @reserved: this field must be 0 + * @dev_token: device token of actual end USB aduio device + * @endian: endianness of this interface + * @service_interval: service interval + **/ +struct q6afe_usb_cfg { + u32 cfg_minor_version; + u32 sample_rate; + u16 bit_width; + u16 num_channels; + u16 data_format; + u16 reserved; + u32 dev_token; + u32 endian; + u32 service_interval; +}; struct q6afe_port_config { struct q6afe_hdmi_cfg hdmi; @@ -212,6 +253,7 @@ 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; @@ -221,6 +263,8 @@ 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, diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c index f67c16fd90b9..39719c3f1767 100644 --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c @@ -81,6 +81,26 @@ 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", @@ -616,6 +636,9 @@ 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 928fd23e2c27..683ae2ae8e50 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -514,6 +514,9 @@ 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) }; @@ -733,6 +736,10 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = { static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { /* Mixer definitions */ + SND_SOC_DAPM_MIXER("USB Mixer", SND_SOC_NOPM, 0, 0, + usb_mixer_controls, + ARRAY_SIZE(usb_mixer_controls)), + SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0, hdmi_mixer_controls, ARRAY_SIZE(hdmi_mixer_controls)), @@ -952,6 +959,7 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + Q6ROUTING_RX_DAPM_ROUTE("USB Mixer", "USB_RX"), Q6ROUTING_RX_DAPM_ROUTE("HDMI Mixer", "HDMI_RX"), Q6ROUTING_RX_DAPM_ROUTE("DISPLAY_PORT_RX Audio Mixer", "DISPLAY_PORT_RX"),
The QC ADSP is able to support USB playback and capture, 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. Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> --- .../sound/qcom,q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 47 +++++ sound/soc/qcom/qdsp6/q6afe.c | 183 ++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 46 ++++- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 8 + 7 files changed, 308 insertions(+), 1 deletion(-)