| Message ID | 20230126031424.14582-10-quic_wcheng@quicinc.com (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | Introduce QC USB SND audio offloading support | expand |
Thanks Wesley for the work, Minor nits. On 26/01/2023 03:14, 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, feedback endpoints > (both implicit and explicit) as well as the isochronous (data) endpoints. > The size of audio samples sent per USB frame (microframe) will be adjusted > based on information received on the feedback endpoint. > > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > sound/soc/qcom/qdsp6/q6afe-dai.c | 48 ++++++ > 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 + > 6 files changed, 308 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c > index 8bb7452b8f18..0773a0882d9b 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; > +} --> This one looks like duplicate of q6slim_hw_params, you could probably use it. > + > 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); indentation. > + break; > default: > return -EINVAL; > } > @@ -495,6 +533,8 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, > } > > static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { > + /* USB playback AFE port receives data for playback, hence use the RX port */ > + {"USB Playback", NULL, "USB_RX"}, Normally we add new entries at the end of this list. > {"HDMI Playback", NULL, "HDMI_RX"}, > {"Display Port Playback", NULL, "DISPLAY_PORT_RX"}, > {"Slimbus Playback", NULL, "SLIMBUS_0_RX"}, > @@ -639,6 +679,12 @@ 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 +749,7 @@ 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), same here. > 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 +1115,7 @@ 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..ca799fc3820e 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}, same > [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); > > +static 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) { > + dev_err(port->afe->dev, "%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)); indentation. > + if (ret) { > + dev_err(port->afe->dev, "%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) { > + 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; this can go in 100 chars line. > + 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); > + 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; same here single line. > + 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; break seems to be missing. > + 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)), > + at the end of the list. > 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"), same. > Q6ROUTING_RX_DAPM_ROUTE("HDMI Mixer", "HDMI_RX"), > Q6ROUTING_RX_DAPM_ROUTE("DISPLAY_PORT_RX Audio Mixer", > "DISPLAY_PORT_RX"), --srini
On 1/25/23 21:14, 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, feedback endpoints > (both implicit and explicit) as well as the isochronous (data) endpoints. > The size of audio samples sent per USB frame (microframe) will be adjusted > based on information received on the feedback endpoint. I think you meant "support all potential endpoint types" It's likely that some USB devices have more endpoints than what the DSP can handle, no? And that brings me back to the question: what is a port and the relationship between port/backend/endpoints? Sorry for being picky on terminology, but if I learned something in days in standardization it's that there shouldn't be any ambiguity on concepts, otherwise everyone is lost at some point. > 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}, And if I look here a port seems to be a very specific AFE concept related to interface type? Do we even need to refer to a port in the USB parts?
Hi Pierre, On 1/26/2023 7:38 AM, Pierre-Louis Bossart wrote: > > > On 1/25/23 21:14, 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, feedback endpoints >> (both implicit and explicit) as well as the isochronous (data) endpoints. >> The size of audio samples sent per USB frame (microframe) will be adjusted >> based on information received on the feedback endpoint. > > I think you meant "support all potential endpoint types" > > It's likely that some USB devices have more endpoints than what the DSP > can handle, no? > True, as we discussed before, we only handle the endpoints for the audio interface. Other endpoints, such as HID, or control is still handled by the main processor. > And that brings me back to the question: what is a port and the > relationship between port/backend/endpoints? > > Sorry for being picky on terminology, but if I learned something in days > in standardization it's that there shouldn't be any ambiguity on > concepts, otherwise everyone is lost at some point. > No worries, I can understand where you're coming from :). After re-reading some of the notations used, I can see where people may be confused. > >> 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}, > > And if I look here a port seems to be a very specific AFE concept > related to interface type? Do we even need to refer to a port in the USB > parts? > Well, this is a design specific to how the Q6 AFE is implemented. There is a concept for an AFE port to be opened. However, as mentioned earlier, the "port" term used in soc-usb should be more for how many USB devices can be supported. If there was a case the audio DSP would support more than one USB device, I believe another AFE port would need to be added. Thanks Wesley Cheng
On 1/30/23 16:54, Wesley Cheng wrote: > Hi Pierre, > > On 1/26/2023 7:38 AM, Pierre-Louis Bossart wrote: >> >> >> On 1/25/23 21:14, 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, feedback endpoints >>> (both implicit and explicit) as well as the isochronous (data) >>> endpoints. >>> The size of audio samples sent per USB frame (microframe) will be >>> adjusted >>> based on information received on the feedback endpoint. >> >> I think you meant "support all potential endpoint types" >> >> It's likely that some USB devices have more endpoints than what the DSP >> can handle, no? >> > > True, as we discussed before, we only handle the endpoints for the audio > interface. Other endpoints, such as HID, or control is still handled by > the main processor. The number of isoc/audio endpoints can be larger than 1 per direction, it's not uncommon for a USB device to have multiple connectors on the front side for instruments, mics, monitor speakers, you name it. Just google 'motu' or 'rme usb' and you'll see examples of USB devices that are very different from plain vanilla headsets. >> And that brings me back to the question: what is a port and the >> relationship between port/backend/endpoints? >> >> Sorry for being picky on terminology, but if I learned something in days >> in standardization it's that there shouldn't be any ambiguity on >> concepts, otherwise everyone is lost at some point. >> > > No worries, I can understand where you're coming from :). After > re-reading some of the notations used, I can see where people may be > confused. > >> >>> 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}, >> >> And if I look here a port seems to be a very specific AFE concept >> related to interface type? Do we even need to refer to a port in the USB >> parts? >> > > Well, this is a design specific to how the Q6 AFE is implemented. There > is a concept for an AFE port to be opened. However, as mentioned > earlier, the "port" term used in soc-usb should be more for how many USB > devices can be supported. > > If there was a case the audio DSP would support more than one USB > device, I believe another AFE port would need to be added. would the suggested infrastructure work though, even if the DSP could deal with multiple endpoints on different devices ? You have static mutexes and ops, can that scale to more than one USB device?
Hi Pierre, On 1/30/2023 3:59 PM, Pierre-Louis Bossart wrote: > > > On 1/30/23 16:54, Wesley Cheng wrote: >> Hi Pierre, >> >> On 1/26/2023 7:38 AM, Pierre-Louis Bossart wrote: >>> >>> >>> On 1/25/23 21:14, 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, feedback endpoints >>>> (both implicit and explicit) as well as the isochronous (data) >>>> endpoints. >>>> The size of audio samples sent per USB frame (microframe) will be >>>> adjusted >>>> based on information received on the feedback endpoint. >>> >>> I think you meant "support all potential endpoint types" >>> >>> It's likely that some USB devices have more endpoints than what the DSP >>> can handle, no? >>> >> >> True, as we discussed before, we only handle the endpoints for the audio >> interface. Other endpoints, such as HID, or control is still handled by >> the main processor. > > The number of isoc/audio endpoints can be larger than 1 per direction, > it's not uncommon for a USB device to have multiple connectors on the > front side for instruments, mics, monitor speakers, you name it. Just > google 'motu' or 'rme usb' and you'll see examples of USB devices that > are very different from plain vanilla headsets. > Thanks for the reference. I tried to do some research on the RME USB audio devices, and they mentioned that they do have a "class compliant mode," which is for compatibility w/ Linux hosts. I didn't see a vendor specific USB SND driver matching the USB VID/PID either, so I am assuming that it uses the USB SND driver as is.(and that Linux doesn't currently support their vendor specific mode) In that case, the device should conform to the UAC2.0 spec (same statement seen on UAC3.0), which states in Section 4.9.1 Standard AS Interface Descriptor Table 4-26: "4 bNumEndpoints 1 Number Number of endpoints used by this interface (excluding endpoint 0). Must be either 0 (no data endpoint), 1 (data endpoint) or 2 (data and explicit feedback endpoint)." So each audio streaming interface should only have 1 data and potentially 1 feedback. However, this device does expose a large number of channels (I saw up to 18 channels), which the USB backend won't be able to support. I still need to check how ASoC behaves if I pass in a profile that the backend can't support. Maybe in the non-class compliant/vendor based class driver, they have the support for multiple EPs per data interface? I don't have one of these devices on hand, so I can't confirm that. >>> And that brings me back to the question: what is a port and the >>> relationship between port/backend/endpoints? >>> >>> Sorry for being picky on terminology, but if I learned something in days >>> in standardization it's that there shouldn't be any ambiguity on >>> concepts, otherwise everyone is lost at some point. >>> >> >> No worries, I can understand where you're coming from :). After >> re-reading some of the notations used, I can see where people may be >> confused. >> >>> >>>> 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}, >>> >>> And if I look here a port seems to be a very specific AFE concept >>> related to interface type? Do we even need to refer to a port in the USB >>> parts? >>> >> >> Well, this is a design specific to how the Q6 AFE is implemented. There >> is a concept for an AFE port to be opened. However, as mentioned >> earlier, the "port" term used in soc-usb should be more for how many USB >> devices can be supported. >> >> If there was a case the audio DSP would support more than one USB >> device, I believe another AFE port would need to be added. > > > would the suggested infrastructure work though, even if the DSP could > deal with multiple endpoints on different devices ? You have static > mutexes and ops, can that scale to more than one USB device? The mutex is only for registering the card, and ensuring atomic access to the list. I don't see how that would block support for having multiple devices being registered to soc-usb. ops are stored per backend device. Greg did want me to re-look at the soc-usb device management, so I will have to rework some of these things. It would be nice to see if we can get it to work like how the headphone jack works, ie interaction between soc-jack and core/jack.c. Thanks Wesley Cheng
On 1/31/23 20:40, Wesley Cheng wrote: > Hi Pierre, > > On 1/30/2023 3:59 PM, Pierre-Louis Bossart wrote: >> >> >> On 1/30/23 16:54, Wesley Cheng wrote: >>> Hi Pierre, >>> >>> On 1/26/2023 7:38 AM, Pierre-Louis Bossart wrote: >>>> >>>> >>>> On 1/25/23 21:14, 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, feedback >>>>> endpoints >>>>> (both implicit and explicit) as well as the isochronous (data) >>>>> endpoints. >>>>> The size of audio samples sent per USB frame (microframe) will be >>>>> adjusted >>>>> based on information received on the feedback endpoint. >>>> >>>> I think you meant "support all potential endpoint types" >>>> >>>> It's likely that some USB devices have more endpoints than what the DSP >>>> can handle, no? >>>> >>> >>> True, as we discussed before, we only handle the endpoints for the audio >>> interface. Other endpoints, such as HID, or control is still handled by >>> the main processor. >> >> The number of isoc/audio endpoints can be larger than 1 per direction, >> it's not uncommon for a USB device to have multiple connectors on the >> front side for instruments, mics, monitor speakers, you name it. Just >> google 'motu' or 'rme usb' and you'll see examples of USB devices that >> are very different from plain vanilla headsets. >> > > Thanks for the reference. > > I tried to do some research on the RME USB audio devices, and they > mentioned that they do have a "class compliant mode," which is for > compatibility w/ Linux hosts. I didn't see a vendor specific USB SND > driver matching the USB VID/PID either, so I am assuming that it uses > the USB SND driver as is.(and that Linux doesn't currently support their > vendor specific mode) In that case, the device should conform to the > UAC2.0 spec (same statement seen on UAC3.0), which states in Section > 4.9.1 Standard AS Interface Descriptor Table 4-26: > > "4 bNumEndpoints 1 Number Number of endpoints used by this > interface (excluding endpoint 0). Must be > either 0 (no data endpoint), 1 (data > endpoint) or 2 (data and explicit feedback > endpoint)." > > So each audio streaming interface should only have 1 data and > potentially 1 feedback. However, this device does expose a large number > of channels (I saw up to 18 channels), which the USB backend won't be > able to support. I still need to check how ASoC behaves if I pass in a > profile that the backend can't support. > > Maybe in the non-class compliant/vendor based class driver, they have > the support for multiple EPs per data interface? I don't have one of > these devices on hand, so I can't confirm that. Look at Figure 3-1 in the UAC2 spec, it shows it's perfectly legal to have multiple Audio Streaming interfaces - but one Audio Control interface only. The fact that there is a restriction to 1 or 2 endpoints per Audio Streaming interface does not really matter if in the end there are multiple endpoints and concurrent isoc transfers happening to/from the same USB device.
Hi Pierre, On 1/31/2023 7:02 PM, Pierre-Louis Bossart wrote: > > > On 1/31/23 20:40, Wesley Cheng wrote: >> Hi Pierre, >> >> On 1/30/2023 3:59 PM, Pierre-Louis Bossart wrote: >>> >>> >>> On 1/30/23 16:54, Wesley Cheng wrote: >>>> Hi Pierre, >>>> >>>> On 1/26/2023 7:38 AM, Pierre-Louis Bossart wrote: >>>>> >>>>> >>>>> On 1/25/23 21:14, 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, feedback >>>>>> endpoints >>>>>> (both implicit and explicit) as well as the isochronous (data) >>>>>> endpoints. >>>>>> The size of audio samples sent per USB frame (microframe) will be >>>>>> adjusted >>>>>> based on information received on the feedback endpoint. >>>>> >>>>> I think you meant "support all potential endpoint types" >>>>> >>>>> It's likely that some USB devices have more endpoints than what the DSP >>>>> can handle, no? >>>>> >>>> >>>> True, as we discussed before, we only handle the endpoints for the audio >>>> interface. Other endpoints, such as HID, or control is still handled by >>>> the main processor. >>> >>> The number of isoc/audio endpoints can be larger than 1 per direction, >>> it's not uncommon for a USB device to have multiple connectors on the >>> front side for instruments, mics, monitor speakers, you name it. Just >>> google 'motu' or 'rme usb' and you'll see examples of USB devices that >>> are very different from plain vanilla headsets. >>> >> >> Thanks for the reference. >> >> I tried to do some research on the RME USB audio devices, and they >> mentioned that they do have a "class compliant mode," which is for >> compatibility w/ Linux hosts. I didn't see a vendor specific USB SND >> driver matching the USB VID/PID either, so I am assuming that it uses >> the USB SND driver as is.(and that Linux doesn't currently support their >> vendor specific mode) In that case, the device should conform to the >> UAC2.0 spec (same statement seen on UAC3.0), which states in Section >> 4.9.1 Standard AS Interface Descriptor Table 4-26: >> >> "4 bNumEndpoints 1 Number Number of endpoints used by this >> interface (excluding endpoint 0). Must be >> either 0 (no data endpoint), 1 (data >> endpoint) or 2 (data and explicit feedback >> endpoint)." >> >> So each audio streaming interface should only have 1 data and >> potentially 1 feedback. However, this device does expose a large number >> of channels (I saw up to 18 channels), which the USB backend won't be >> able to support. I still need to check how ASoC behaves if I pass in a >> profile that the backend can't support. >> >> Maybe in the non-class compliant/vendor based class driver, they have >> the support for multiple EPs per data interface? I don't have one of >> these devices on hand, so I can't confirm that. > > Look at Figure 3-1 in the UAC2 spec, it shows it's perfectly legal to > have multiple Audio Streaming interfaces - but one Audio Control > interface only. > > The fact that there is a restriction to 1 or 2 endpoints per Audio > Streaming interface does not really matter if in the end there are > multiple endpoints and concurrent isoc transfers happening to/from the > same USB device. So the reason I wanted to mention the max number of EPs within the audio streaming descriptor is because the USB SND driver currently creates streams based off of the number of AS desc: static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) { ... for (i = 0; i < assoc->bInterfaceCount; i++) { int intf = assoc->bFirstInterface + i; if (intf != ctrlif) snd_usb_create_stream(chip, ctrlif, intf); } "assoc" is the audio control interface desc. In the end, when userspace initiates a playback session, it operates on the streams created (which contains at max 1 isoc and 1 feedback ep) In short, the audio DSP doesn't need to consider handling more than 1 isoc ep (and potentially 1 feedback). I believe that each audio stream creates a separate PCM device, so userspace is still free to attempt to activate another audio stream. I believe # of PCM devices created matches the # of streams, so when userspace does activate another session, it would be on an entirely different substream, and can be handled through the USB SND (non-offload) path. If attempted to open the substream used by the offload path, then we would reject is based on the new change. Thanks Wesley Cheng
Hi Pierre, On 2/2/2023 5:23 PM, Wesley Cheng wrote: > Hi Pierre, > > On 1/31/2023 7:02 PM, Pierre-Louis Bossart wrote: >> >> >> On 1/31/23 20:40, Wesley Cheng wrote: >>> Hi Pierre, >>> >>> On 1/30/2023 3:59 PM, Pierre-Louis Bossart wrote: >>>> >>>> >>>> On 1/30/23 16:54, Wesley Cheng wrote: >>>>> Hi Pierre, >>>>> >>>>> On 1/26/2023 7:38 AM, Pierre-Louis Bossart wrote: >>>>>> >>>>>> >>>>>> On 1/25/23 21:14, 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, feedback >>>>>>> endpoints >>>>>>> (both implicit and explicit) as well as the isochronous (data) >>>>>>> endpoints. >>>>>>> The size of audio samples sent per USB frame (microframe) will be >>>>>>> adjusted >>>>>>> based on information received on the feedback endpoint. >>>>>> >>>>>> I think you meant "support all potential endpoint types" >>>>>> >>>>>> It's likely that some USB devices have more endpoints than what >>>>>> the DSP >>>>>> can handle, no? >>>>>> >>>>> >>>>> True, as we discussed before, we only handle the endpoints for the >>>>> audio >>>>> interface. Other endpoints, such as HID, or control is still >>>>> handled by >>>>> the main processor. >>>> >>>> The number of isoc/audio endpoints can be larger than 1 per direction, >>>> it's not uncommon for a USB device to have multiple connectors on the >>>> front side for instruments, mics, monitor speakers, you name it. Just >>>> google 'motu' or 'rme usb' and you'll see examples of USB devices that >>>> are very different from plain vanilla headsets. >>>> >>> >>> Thanks for the reference. >>> >>> I tried to do some research on the RME USB audio devices, and they >>> mentioned that they do have a "class compliant mode," which is for >>> compatibility w/ Linux hosts. I didn't see a vendor specific USB SND >>> driver matching the USB VID/PID either, so I am assuming that it uses >>> the USB SND driver as is.(and that Linux doesn't currently support their >>> vendor specific mode) In that case, the device should conform to the >>> UAC2.0 spec (same statement seen on UAC3.0), which states in Section >>> 4.9.1 Standard AS Interface Descriptor Table 4-26: >>> >>> "4 bNumEndpoints 1 Number Number of endpoints used by this >>> interface (excluding endpoint 0). Must be >>> either 0 (no data endpoint), 1 (data >>> endpoint) or 2 (data and explicit feedback >>> endpoint)." >>> >>> So each audio streaming interface should only have 1 data and >>> potentially 1 feedback. However, this device does expose a large number >>> of channels (I saw up to 18 channels), which the USB backend won't be >>> able to support. I still need to check how ASoC behaves if I pass in a >>> profile that the backend can't support. Getting back to passing in a format/profile that the USB BE doesn't support. It looks like ASoC doesn't actually check against the PCM HW params received (for components), so the audio playback does still occur even though its outside of what we support. Will need to add changes to specifically check for # of channels, format, etc... before we allow the session to proceed. Thanks Wesley Cheng
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 8bb7452b8f18..0773a0882d9b 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 +533,8 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, } static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { + /* USB playback AFE port receives data for playback, hence use the RX port */ + {"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 +679,12 @@ 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 +749,7 @@ 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 +1115,7 @@ 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..ca799fc3820e 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); +static 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) { + dev_err(port->afe->dev, "%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) { + dev_err(port->afe->dev, "%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) { + 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); + 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 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, feedback endpoints (both implicit and explicit) as well as the isochronous (data) endpoints. The size of audio samples sent per USB frame (microframe) will be adjusted based on information received on the feedback endpoint. Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 48 ++++++ 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 + 6 files changed, 308 insertions(+), 1 deletion(-)