| Message ID | 20230829210657.9904-1-quic_wcheng@quicinc.com |
|---|---|
| Headers | show |
| Series | Introduce QC USB SND audio offloading support | expand |
On 8/29/2023 11:06 PM, Wesley Cheng wrote: > Create a USB BE component that will register a new USB port to the ASoC USB > framework. This will handle determination on if the requested audio > profile is supported by the USB device currently selected. > > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > include/sound/q6usboffload.h | 20 ++++ > sound/soc/qcom/Kconfig | 4 + > sound/soc/qcom/qdsp6/Makefile | 1 + > sound/soc/qcom/qdsp6/q6usb.c | 200 ++++++++++++++++++++++++++++++++++ > 4 files changed, 225 insertions(+) > create mode 100644 include/sound/q6usboffload.h > create mode 100644 sound/soc/qcom/qdsp6/q6usb.c > > diff --git a/include/sound/q6usboffload.h b/include/sound/q6usboffload.h > new file mode 100644 > index 000000000000..4fb1912d9f55 > --- /dev/null > +++ b/include/sound/q6usboffload.h > @@ -0,0 +1,20 @@ > +/* SPDX-License-Identifier: GPL-2.0 > + * > + * linux/sound/q6usboffload.h -- QDSP6 USB offload > + * > + * Copyright (c) 2022-2023 Qualcomm Innovation Center, Inc. All rights reserved. > + */ > + > +/** > + * struct q6usb_offload > + * @dev - dev handle to usb be > + * @sid - streamID for iommu > + * @intr_num - usb interrupter number > + * @domain - allocated iommu domain > + **/ > +struct q6usb_offload { > + struct device *dev; > + long long sid; > + u32 intr_num; > + struct iommu_domain *domain; > +}; > diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig > index e7b00d1d9e99..bb285af6bb04 100644 > --- a/sound/soc/qcom/Kconfig > +++ b/sound/soc/qcom/Kconfig > @@ -114,6 +114,9 @@ config SND_SOC_QDSP6_APM > config SND_SOC_QDSP6_PRM_LPASS_CLOCKS > tristate > > +config SND_SOC_QDSP6_USB > + tristate > + > config SND_SOC_QDSP6_PRM > tristate > select SND_SOC_QDSP6_PRM_LPASS_CLOCKS > @@ -134,6 +137,7 @@ config SND_SOC_QDSP6 > select SND_SOC_TOPOLOGY > select SND_SOC_QDSP6_APM > select SND_SOC_QDSP6_PRM > + select SND_SOC_QDSP6_USB > help > To add support for MSM QDSP6 Soc Audio. > This will enable sound soc platform specific > diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile > index 3963bf234664..c9457ee898d0 100644 > --- a/sound/soc/qcom/qdsp6/Makefile > +++ b/sound/soc/qcom/qdsp6/Makefile > @@ -17,3 +17,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_APM_DAI) += q6apm-dai.o > obj-$(CONFIG_SND_SOC_QDSP6_APM_LPASS_DAI) += q6apm-lpass-dais.o > obj-$(CONFIG_SND_SOC_QDSP6_PRM) += q6prm.o > obj-$(CONFIG_SND_SOC_QDSP6_PRM_LPASS_CLOCKS) += q6prm-clocks.o > +obj-$(CONFIG_SND_SOC_QDSP6_USB) += q6usb.o > diff --git a/sound/soc/qcom/qdsp6/q6usb.c b/sound/soc/qcom/qdsp6/q6usb.c > new file mode 100644 > index 000000000000..88aa0a64201a > --- /dev/null > +++ b/sound/soc/qcom/qdsp6/q6usb.c > @@ -0,0 +1,200 @@ > +// SPDX-License-Identifier: GPL-2.0 > +/* > + * Copyright (c) 2022-2023 Qualcomm Innovation Center, Inc. All rights reserved. > + */ > + > +#include <linux/err.h> > +#include <linux/init.h> > +#include <linux/module.h> > +#include <linux/device.h> > +#include <linux/platform_device.h> > +#include <linux/slab.h> > +#include <linux/iommu.h> > +#include <linux/dma-mapping.h> > +#include <linux/dma-map-ops.h> > + > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include <sound/soc-usb.h> > +#include <sound/pcm_params.h> > +#include <sound/asound.h> > +#include <sound/q6usboffload.h> > + > +#include "q6dsp-lpass-ports.h" > +#include "q6afe.h" > + > +#define SID_MASK 0xF > + > +struct q6usb_port_data { > + struct q6afe_usb_cfg usb_cfg; > + struct snd_soc_usb *usb; > + struct q6usb_offload priv; > + int active_idx; > +}; > + > +static const struct snd_soc_dapm_widget q6usb_dai_widgets[] = { > + SND_SOC_DAPM_HP("USB_RX_BE", NULL), > +}; > + > +static const struct snd_soc_dapm_route q6usb_dapm_routes[] = { > + {"USB Playback", NULL, "USB_RX_BE"}, > +}; > + > +static int q6usb_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + return 0; > +} Missing new line after closing bracket between function and following struct. > +static const struct snd_soc_dai_ops q6usb_ops = { > + .hw_params = q6usb_hw_params, > +}; > + > +static struct snd_soc_dai_driver q6usb_be_dais[] = { > + { > + .playback = { > + .stream_name = "USB BE RX", > + .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_max = 192000, > + .rate_min = 8000, > + }, > + .id = USB_RX, > + .name = "USB_RX_BE", > + .ops = &q6usb_ops, > + }, > +}; > + > +static int q6usb_audio_ports_of_xlate_dai_name(struct snd_soc_component *component, > + const struct of_phandle_args *args, > + const char **dai_name) > +{ > + int id = args->args[0]; > + int ret = -EINVAL; > + int i; > + > + for (i = 0; i < ARRAY_SIZE(q6usb_be_dais); i++) { Double space after second i. > + if (q6usb_be_dais[i].id == id) { > + *dai_name = q6usb_be_dais[i].name; > + ret = 0; > + break; > + } > + } > + > + return ret; > +} > + > +static int q6usb_alsa_connection_cb(struct snd_soc_usb *usb, > + struct snd_soc_usb_device *sdev, bool connected) > +{ > + struct q6usb_port_data *data; > + > + if (!usb->component) > + return -ENODEV; > + > + data = dev_get_drvdata(usb->component->dev); > + > + if (connected) > + /* We only track the latest USB headset plugged in */ > + data->active_idx = sdev->card_idx; Maybe add brackets around both comment and code? Not sure what guidance there is in such cases, but above code looks weird to me. > + > + return 0; > +} > + > +static int q6usb_component_probe(struct snd_soc_component *component) > +{ > + struct q6usb_port_data *data = dev_get_drvdata(component->dev); > + > + data->usb = snd_soc_usb_add_port(component->dev, &data->priv, q6usb_alsa_connection_cb); > + if (IS_ERR(data->usb)) { > + dev_err(component->dev, "failed to add usb port\n"); > + return -ENODEV; > + } > + > + data->usb->component = component; > + > + return 0; > +} > + > +static void q6usb_component_remove(struct snd_soc_component *component) > +{ > + snd_soc_usb_remove_port(component->dev); > +} > + > +static const struct snd_soc_component_driver q6usb_dai_component = { > + .probe = q6usb_component_probe, > + .remove = q6usb_component_remove, > + .name = "q6usb-dai-component", > + .dapm_widgets = q6usb_dai_widgets, > + .num_dapm_widgets = ARRAY_SIZE(q6usb_dai_widgets), > + .dapm_routes = q6usb_dapm_routes, > + .num_dapm_routes = ARRAY_SIZE(q6usb_dapm_routes), > + .of_xlate_dai_name = q6usb_audio_ports_of_xlate_dai_name, > +}; > + > +static int q6usb_dai_dev_probe(struct platform_device *pdev) > +{ > + struct device_node *node = pdev->dev.of_node; > + struct q6usb_port_data *data; > + struct device *dev = &pdev->dev; > + struct of_phandle_args args; > + int ret; > + > + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); > + if (!data) > + return -ENOMEM; > + > + ret = of_property_read_u32(node, "qcom,usb-audio-intr-num", > + &data->priv.intr_num); > + if (ret) { > + dev_err(&pdev->dev, "failed to read intr num.\n"); > + return ret; > + } > + > + ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args); > + if (ret < 0) > + data->priv.sid = -1; > + else > + data->priv.sid = args.args[0] & SID_MASK; > + > + data->priv.domain = iommu_get_domain_for_dev(&pdev->dev); > + > + data->priv.dev = dev; > + dev_set_drvdata(dev, data); > + > + return devm_snd_soc_register_component(dev, &q6usb_dai_component, > + q6usb_be_dais, ARRAY_SIZE(q6usb_be_dais)); > +} > + > +static int q6usb_dai_dev_remove(struct platform_device *pdev) > +{ > + return 0; > +} Does platform driver really need empty remove function? Remove it. > + > +static const struct of_device_id q6usb_dai_device_id[] = { > + { .compatible = "qcom,q6usb" }, > + {}, > +}; > +MODULE_DEVICE_TABLE(of, q6usb_dai_device_id); > + > +static struct platform_driver q6usb_dai_platform_driver = { > + .driver = { > + .name = "q6usb-dai", > + .of_match_table = of_match_ptr(q6usb_dai_device_id), > + }, > + .probe = q6usb_dai_dev_probe, > + .remove = q6usb_dai_dev_remove, > +}; > +module_platform_driver(q6usb_dai_platform_driver); > + > +MODULE_DESCRIPTION("Q6 USB backend dai driver"); > +MODULE_LICENSE("GPL");
On 8/29/2023 11:06 PM, Wesley Cheng wrote: > Some vendor modules will utilize useful parsing and endpoint management > APIs to start audio playback/capture. > > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > sound/usb/card.c | 4 +++ > sound/usb/endpoint.c | 1 + > sound/usb/helper.c | 1 + > sound/usb/pcm.c | 67 +++++++++++++++++++++++++++++++++----------- > sound/usb/pcm.h | 11 ++++++++ > 5 files changed, 67 insertions(+), 17 deletions(-) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 067a1e82f4bf..b45b6daee7b7 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -1053,6 +1053,7 @@ int snd_usb_lock_shutdown(struct snd_usb_audio *chip) > wake_up(&chip->shutdown_wait); > return err; > } > +EXPORT_SYMBOL_GPL(snd_usb_lock_shutdown); > > /* autosuspend and unlock the shutdown */ > void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) > @@ -1061,6 +1062,7 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) > if (atomic_dec_and_test(&chip->usage_count)) > wake_up(&chip->shutdown_wait); > } > +EXPORT_SYMBOL_GPL(snd_usb_unlock_shutdown); > > int snd_usb_autoresume(struct snd_usb_audio *chip) > { > @@ -1083,6 +1085,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) > } > return 0; > } > +EXPORT_SYMBOL_GPL(snd_usb_autoresume); > > void snd_usb_autosuspend(struct snd_usb_audio *chip) > { > @@ -1096,6 +1099,7 @@ void snd_usb_autosuspend(struct snd_usb_audio *chip) > for (i = 0; i < chip->num_interfaces; i++) > usb_autopm_put_interface(chip->intf[i]); > } > +EXPORT_SYMBOL_GPL(snd_usb_autosuspend); > > static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) > { > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index a385e85c4650..aac92e0b8aa2 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -1503,6 +1503,7 @@ int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, > mutex_unlock(&chip->mutex); > return err; > } > +EXPORT_SYMBOL_GPL(snd_usb_endpoint_prepare); > > /* get the current rate set to the given clock by any endpoint */ > int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock) > diff --git a/sound/usb/helper.c b/sound/usb/helper.c > index bf80e55d013a..4322ae3738e6 100644 > --- a/sound/usb/helper.c > +++ b/sound/usb/helper.c > @@ -62,6 +62,7 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype > } > return NULL; > } > +EXPORT_SYMBOL_GPL(snd_usb_find_csint_desc); > > /* > * Wrapper for usb_control_msg(). > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index 08bf535ed163..999f66080649 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -148,6 +148,16 @@ find_format(struct list_head *fmt_list_head, snd_pcm_format_t format, > return found; > } > > +const struct audioformat * > +snd_usb_find_format(struct list_head *fmt_list_head, snd_pcm_format_t format, > + unsigned int rate, unsigned int channels, bool strict_match, > + struct snd_usb_substream *subs) > +{ > + return find_format(fmt_list_head, format, rate, channels, strict_match, > + subs); > +} > +EXPORT_SYMBOL_GPL(snd_usb_find_format); > + > static const struct audioformat * > find_substream_format(struct snd_usb_substream *subs, > const struct snd_pcm_hw_params *params) > @@ -157,6 +167,14 @@ find_substream_format(struct snd_usb_substream *subs, > true, subs); > } > > +const struct audioformat * > +snd_usb_find_substream_format(struct snd_usb_substream *subs, > + const struct snd_pcm_hw_params *params) > +{ > + return find_substream_format(subs, params); > +} > +EXPORT_SYMBOL_GPL(snd_usb_find_substream_format); > + > bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs) > { > const struct audioformat *fp; > @@ -461,20 +479,9 @@ static void close_endpoints(struct snd_usb_audio *chip, > } > } > > -/* > - * hw_params callback > - * > - * allocate a buffer and set the given audio format. > - * > - * so far we use a physically linear buffer although packetize transfer > - * doesn't need a continuous area. > - * if sg buffer is supported on the later version of alsa, we'll follow > - * that. > - */ > -static int snd_usb_hw_params(struct snd_pcm_substream *substream, > - struct snd_pcm_hw_params *hw_params) > +int snd_usb_attach_endpoints(struct snd_usb_substream *subs, > + struct snd_pcm_hw_params *hw_params) > { > - struct snd_usb_substream *subs = substream->runtime->private_data; > struct snd_usb_audio *chip = subs->stream->chip; > const struct audioformat *fmt; > const struct audioformat *sync_fmt; > @@ -499,7 +506,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, > if (fmt->implicit_fb) { > sync_fmt = snd_usb_find_implicit_fb_sync_format(chip, fmt, > hw_params, > - !substream->stream, > + !subs->direction, > &sync_fixed_rate); > if (!sync_fmt) { > usb_audio_dbg(chip, > @@ -579,15 +586,28 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, > > return ret; > } > +EXPORT_SYMBOL_GPL(snd_usb_attach_endpoints); > > /* > - * hw_free callback > + * hw_params callback > * > - * reset the audio format and release the buffer > + * allocate a buffer and set the given audio format. > + * > + * so far we use a physically linear buffer although packetize transfer > + * doesn't need a continuous area. > + * if sg buffer is supported on the later version of alsa, we'll follow > + * that. > */ > -static int snd_usb_hw_free(struct snd_pcm_substream *substream) > +static int snd_usb_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *hw_params) > { > struct snd_usb_substream *subs = substream->runtime->private_data; > + > + return snd_usb_attach_endpoints(subs, hw_params); > +} > + > +int snd_usb_detach_endpoint(struct snd_usb_substream *subs) > +{ > struct snd_usb_audio *chip = subs->stream->chip; > > snd_media_stop_pipeline(subs); > @@ -603,6 +623,19 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) > > return 0; > } > +EXPORT_SYMBOL_GPL(snd_usb_detach_endpoint); > + > +/* > + * hw_free callback > + * > + * reset the audio format and release the buffer > + */ > +static int snd_usb_hw_free(struct snd_pcm_substream *substream) > +{ > + struct snd_usb_substream *subs = substream->runtime->private_data; > + > + return snd_usb_detach_endpoint(subs); > +} > > /* free-wheeling mode? (e.g. dmix) */ > static int in_free_wheeling_mode(struct snd_pcm_runtime *runtime) > diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h > index 388fe2ba346d..e36df3611a05 100644 > --- a/sound/usb/pcm.h > +++ b/sound/usb/pcm.h > @@ -15,4 +15,15 @@ void snd_usb_preallocate_buffer(struct snd_usb_substream *subs); > int snd_usb_audioformat_set_sync_ep(struct snd_usb_audio *chip, > struct audioformat *fmt); > > +const struct audioformat * > +snd_usb_find_format(struct list_head *fmt_list_head, snd_pcm_format_t format, > + unsigned int rate, unsigned int channels, bool strict_match, > + struct snd_usb_substream *subs); > +const struct audioformat * > +snd_usb_find_substream_format(struct snd_usb_substream *subs, > + const struct snd_pcm_hw_params *params); > + > +int snd_usb_attach_endpoints(struct snd_usb_substream *subs, > + struct snd_pcm_hw_params *hw_params); > +int snd_usb_detach_endpoint(struct snd_usb_substream *subs); > #endif /* __USBAUDIO_PCM_H */ Why is it multiple "endpoints" when attaching, but only one "endpoint" when detaching? Both seem to be getting similar arguments.
Hi, On 8/30/2023 5:50 AM, Amadeusz Sławiński wrote: > On 8/29/2023 11:06 PM, Wesley Cheng wrote: >> Create a USB BE component that will register a new USB port to the >> ASoC USB >> framework. This will handle determination on if the requested audio >> profile is supported by the USB device currently selected. >> >> Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> >> --- >> include/sound/q6usboffload.h | 20 ++++ >> sound/soc/qcom/Kconfig | 4 + >> sound/soc/qcom/qdsp6/Makefile | 1 + >> sound/soc/qcom/qdsp6/q6usb.c | 200 ++++++++++++++++++++++++++++++++++ >> 4 files changed, 225 insertions(+) >> create mode 100644 include/sound/q6usboffload.h >> create mode 100644 sound/soc/qcom/qdsp6/q6usb.c >> >> diff --git a/include/sound/q6usboffload.h b/include/sound/q6usboffload.h >> new file mode 100644 >> index 000000000000..4fb1912d9f55 >> --- /dev/null >> +++ b/include/sound/q6usboffload.h >> @@ -0,0 +1,20 @@ >> +/* SPDX-License-Identifier: GPL-2.0 >> + * >> + * linux/sound/q6usboffload.h -- QDSP6 USB offload >> + * >> + * Copyright (c) 2022-2023 Qualcomm Innovation Center, Inc. All >> rights reserved. >> + */ >> + >> +/** >> + * struct q6usb_offload >> + * @dev - dev handle to usb be >> + * @sid - streamID for iommu >> + * @intr_num - usb interrupter number >> + * @domain - allocated iommu domain >> + **/ >> +struct q6usb_offload { >> + struct device *dev; >> + long long sid; >> + u32 intr_num; >> + struct iommu_domain *domain; >> +}; >> diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig >> index e7b00d1d9e99..bb285af6bb04 100644 >> --- a/sound/soc/qcom/Kconfig >> +++ b/sound/soc/qcom/Kconfig >> @@ -114,6 +114,9 @@ config SND_SOC_QDSP6_APM >> config SND_SOC_QDSP6_PRM_LPASS_CLOCKS >> tristate >> +config SND_SOC_QDSP6_USB >> + tristate >> + >> config SND_SOC_QDSP6_PRM >> tristate >> select SND_SOC_QDSP6_PRM_LPASS_CLOCKS >> @@ -134,6 +137,7 @@ config SND_SOC_QDSP6 >> select SND_SOC_TOPOLOGY >> select SND_SOC_QDSP6_APM >> select SND_SOC_QDSP6_PRM >> + select SND_SOC_QDSP6_USB >> help >> To add support for MSM QDSP6 Soc Audio. >> This will enable sound soc platform specific >> diff --git a/sound/soc/qcom/qdsp6/Makefile >> b/sound/soc/qcom/qdsp6/Makefile >> index 3963bf234664..c9457ee898d0 100644 >> --- a/sound/soc/qcom/qdsp6/Makefile >> +++ b/sound/soc/qcom/qdsp6/Makefile >> @@ -17,3 +17,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_APM_DAI) += q6apm-dai.o >> obj-$(CONFIG_SND_SOC_QDSP6_APM_LPASS_DAI) += q6apm-lpass-dais.o >> obj-$(CONFIG_SND_SOC_QDSP6_PRM) += q6prm.o >> obj-$(CONFIG_SND_SOC_QDSP6_PRM_LPASS_CLOCKS) += q6prm-clocks.o >> +obj-$(CONFIG_SND_SOC_QDSP6_USB) += q6usb.o >> diff --git a/sound/soc/qcom/qdsp6/q6usb.c b/sound/soc/qcom/qdsp6/q6usb.c >> new file mode 100644 >> index 000000000000..88aa0a64201a >> --- /dev/null >> +++ b/sound/soc/qcom/qdsp6/q6usb.c >> @@ -0,0 +1,200 @@ >> +// SPDX-License-Identifier: GPL-2.0 >> +/* >> + * Copyright (c) 2022-2023 Qualcomm Innovation Center, Inc. All >> rights reserved. >> + */ >> + >> +#include <linux/err.h> >> +#include <linux/init.h> >> +#include <linux/module.h> >> +#include <linux/device.h> >> +#include <linux/platform_device.h> >> +#include <linux/slab.h> >> +#include <linux/iommu.h> >> +#include <linux/dma-mapping.h> >> +#include <linux/dma-map-ops.h> >> + >> +#include <sound/pcm.h> >> +#include <sound/soc.h> >> +#include <sound/soc-usb.h> >> +#include <sound/pcm_params.h> >> +#include <sound/asound.h> >> +#include <sound/q6usboffload.h> >> + >> +#include "q6dsp-lpass-ports.h" >> +#include "q6afe.h" >> + >> +#define SID_MASK 0xF >> + >> +struct q6usb_port_data { >> + struct q6afe_usb_cfg usb_cfg; >> + struct snd_soc_usb *usb; >> + struct q6usb_offload priv; >> + int active_idx; >> +}; >> + >> +static const struct snd_soc_dapm_widget q6usb_dai_widgets[] = { >> + SND_SOC_DAPM_HP("USB_RX_BE", NULL), >> +}; >> + >> +static const struct snd_soc_dapm_route q6usb_dapm_routes[] = { >> + {"USB Playback", NULL, "USB_RX_BE"}, >> +}; >> + >> +static int q6usb_hw_params(struct snd_pcm_substream *substream, >> + struct snd_pcm_hw_params *params, >> + struct snd_soc_dai *dai) >> +{ >> + return 0; >> +} > > Missing new line after closing bracket between function and following > struct. > Thanks for the review, will fix this. >> +static const struct snd_soc_dai_ops q6usb_ops = { >> + .hw_params = q6usb_hw_params, >> +}; >> + >> +static struct snd_soc_dai_driver q6usb_be_dais[] = { >> + { >> + .playback = { >> + .stream_name = "USB BE RX", >> + .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_max = 192000, >> + .rate_min = 8000, >> + }, >> + .id = USB_RX, >> + .name = "USB_RX_BE", >> + .ops = &q6usb_ops, >> + }, >> +}; >> + >> +static int q6usb_audio_ports_of_xlate_dai_name(struct >> snd_soc_component *component, >> + const struct of_phandle_args *args, >> + const char **dai_name) >> +{ >> + int id = args->args[0]; >> + int ret = -EINVAL; >> + int i; >> + >> + for (i = 0; i < ARRAY_SIZE(q6usb_be_dais); i++) { > > Double space after second i. > Will change. >> + if (q6usb_be_dais[i].id == id) { >> + *dai_name = q6usb_be_dais[i].name; >> + ret = 0; >> + break; >> + } >> + } >> + >> + return ret; >> +} >> + >> +static int q6usb_alsa_connection_cb(struct snd_soc_usb *usb, >> + struct snd_soc_usb_device *sdev, bool connected) >> +{ >> + struct q6usb_port_data *data; >> + >> + if (!usb->component) >> + return -ENODEV; >> + >> + data = dev_get_drvdata(usb->component->dev); >> + >> + if (connected) >> + /* We only track the latest USB headset plugged in */ >> + data->active_idx = sdev->card_idx; > > Maybe add brackets around both comment and code? Not sure what guidance > there is in such cases, but above code looks weird to me. > Sure. >> + >> + return 0; >> +} >> + >> +static int q6usb_component_probe(struct snd_soc_component *component) >> +{ >> + struct q6usb_port_data *data = dev_get_drvdata(component->dev); >> + >> + data->usb = snd_soc_usb_add_port(component->dev, &data->priv, >> q6usb_alsa_connection_cb); >> + if (IS_ERR(data->usb)) { >> + dev_err(component->dev, "failed to add usb port\n"); >> + return -ENODEV; >> + } >> + >> + data->usb->component = component; >> + >> + return 0; >> +} >> + >> +static void q6usb_component_remove(struct snd_soc_component *component) >> +{ >> + snd_soc_usb_remove_port(component->dev); >> +} >> + >> +static const struct snd_soc_component_driver q6usb_dai_component = { >> + .probe = q6usb_component_probe, >> + .remove = q6usb_component_remove, >> + .name = "q6usb-dai-component", >> + .dapm_widgets = q6usb_dai_widgets, >> + .num_dapm_widgets = ARRAY_SIZE(q6usb_dai_widgets), >> + .dapm_routes = q6usb_dapm_routes, >> + .num_dapm_routes = ARRAY_SIZE(q6usb_dapm_routes), >> + .of_xlate_dai_name = q6usb_audio_ports_of_xlate_dai_name, >> +}; >> + >> +static int q6usb_dai_dev_probe(struct platform_device *pdev) >> +{ >> + struct device_node *node = pdev->dev.of_node; >> + struct q6usb_port_data *data; >> + struct device *dev = &pdev->dev; >> + struct of_phandle_args args; >> + int ret; >> + >> + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); >> + if (!data) >> + return -ENOMEM; >> + >> + ret = of_property_read_u32(node, "qcom,usb-audio-intr-num", >> + &data->priv.intr_num); >> + if (ret) { >> + dev_err(&pdev->dev, "failed to read intr num.\n"); >> + return ret; >> + } >> + >> + ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args); >> + if (ret < 0) >> + data->priv.sid = -1; >> + else >> + data->priv.sid = args.args[0] & SID_MASK; >> + >> + data->priv.domain = iommu_get_domain_for_dev(&pdev->dev); >> + >> + data->priv.dev = dev; >> + dev_set_drvdata(dev, data); >> + >> + return devm_snd_soc_register_component(dev, &q6usb_dai_component, >> + q6usb_be_dais, ARRAY_SIZE(q6usb_be_dais)); >> +} >> + >> +static int q6usb_dai_dev_remove(struct platform_device *pdev) >> +{ >> + return 0; >> +} > > Does platform driver really need empty remove function? Remove it. > Wasn't too sure about this either, so I included it to be consistent. Will remove this and add a small comment on why it isn't required.. Thanks Wesley Cheng
Hi Amadeusz, On 8/30/2023 5:50 AM, Amadeusz Sławiński wrote: > On 8/29/2023 11:06 PM, Wesley Cheng wrote: >> Some vendor modules will utilize useful parsing and endpoint management >> APIs to start audio playback/capture. >> >> Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> >> --- >> sound/usb/card.c | 4 +++ >> sound/usb/endpoint.c | 1 + >> sound/usb/helper.c | 1 + >> sound/usb/pcm.c | 67 +++++++++++++++++++++++++++++++++----------- >> sound/usb/pcm.h | 11 ++++++++ >> 5 files changed, 67 insertions(+), 17 deletions(-) >> >> diff --git a/sound/usb/card.c b/sound/usb/card.c >> index 067a1e82f4bf..b45b6daee7b7 100644 >> --- a/sound/usb/card.c >> +++ b/sound/usb/card.c >> @@ -1053,6 +1053,7 @@ int snd_usb_lock_shutdown(struct snd_usb_audio >> *chip) >> wake_up(&chip->shutdown_wait); >> return err; >> } >> +EXPORT_SYMBOL_GPL(snd_usb_lock_shutdown); >> /* autosuspend and unlock the shutdown */ >> void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) >> @@ -1061,6 +1062,7 @@ void snd_usb_unlock_shutdown(struct >> snd_usb_audio *chip) >> if (atomic_dec_and_test(&chip->usage_count)) >> wake_up(&chip->shutdown_wait); >> } >> +EXPORT_SYMBOL_GPL(snd_usb_unlock_shutdown); >> int snd_usb_autoresume(struct snd_usb_audio *chip) >> { >> @@ -1083,6 +1085,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) >> } >> return 0; >> } >> +EXPORT_SYMBOL_GPL(snd_usb_autoresume); >> void snd_usb_autosuspend(struct snd_usb_audio *chip) >> { >> @@ -1096,6 +1099,7 @@ void snd_usb_autosuspend(struct snd_usb_audio >> *chip) >> for (i = 0; i < chip->num_interfaces; i++) >> usb_autopm_put_interface(chip->intf[i]); >> } >> +EXPORT_SYMBOL_GPL(snd_usb_autosuspend); >> static int usb_audio_suspend(struct usb_interface *intf, >> pm_message_t message) >> { >> diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c >> index a385e85c4650..aac92e0b8aa2 100644 >> --- a/sound/usb/endpoint.c >> +++ b/sound/usb/endpoint.c >> @@ -1503,6 +1503,7 @@ int snd_usb_endpoint_prepare(struct >> snd_usb_audio *chip, >> mutex_unlock(&chip->mutex); >> return err; >> } >> +EXPORT_SYMBOL_GPL(snd_usb_endpoint_prepare); >> /* get the current rate set to the given clock by any endpoint */ >> int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int >> clock) >> diff --git a/sound/usb/helper.c b/sound/usb/helper.c >> index bf80e55d013a..4322ae3738e6 100644 >> --- a/sound/usb/helper.c >> +++ b/sound/usb/helper.c >> @@ -62,6 +62,7 @@ void *snd_usb_find_csint_desc(void *buffer, int >> buflen, void *after, u8 dsubtype >> } >> return NULL; >> } >> +EXPORT_SYMBOL_GPL(snd_usb_find_csint_desc); >> /* >> * Wrapper for usb_control_msg(). >> diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c >> index 08bf535ed163..999f66080649 100644 >> --- a/sound/usb/pcm.c >> +++ b/sound/usb/pcm.c >> @@ -148,6 +148,16 @@ find_format(struct list_head *fmt_list_head, >> snd_pcm_format_t format, >> return found; >> } >> +const struct audioformat * >> +snd_usb_find_format(struct list_head *fmt_list_head, snd_pcm_format_t >> format, >> + unsigned int rate, unsigned int channels, bool strict_match, >> + struct snd_usb_substream *subs) >> +{ >> + return find_format(fmt_list_head, format, rate, channels, >> strict_match, >> + subs); >> +} >> +EXPORT_SYMBOL_GPL(snd_usb_find_format); >> + >> static const struct audioformat * >> find_substream_format(struct snd_usb_substream *subs, >> const struct snd_pcm_hw_params *params) >> @@ -157,6 +167,14 @@ find_substream_format(struct snd_usb_substream >> *subs, >> true, subs); >> } >> +const struct audioformat * >> +snd_usb_find_substream_format(struct snd_usb_substream *subs, >> + const struct snd_pcm_hw_params *params) >> +{ >> + return find_substream_format(subs, params); >> +} >> +EXPORT_SYMBOL_GPL(snd_usb_find_substream_format); >> + >> bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs) >> { >> const struct audioformat *fp; >> @@ -461,20 +479,9 @@ static void close_endpoints(struct snd_usb_audio >> *chip, >> } >> } >> -/* >> - * hw_params callback >> - * >> - * allocate a buffer and set the given audio format. >> - * >> - * so far we use a physically linear buffer although packetize transfer >> - * doesn't need a continuous area. >> - * if sg buffer is supported on the later version of alsa, we'll follow >> - * that. >> - */ >> -static int snd_usb_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *hw_params) >> +int snd_usb_attach_endpoints(struct snd_usb_substream *subs, >> + struct snd_pcm_hw_params *hw_params) >> { >> - struct snd_usb_substream *subs = substream->runtime->private_data; >> struct snd_usb_audio *chip = subs->stream->chip; >> const struct audioformat *fmt; >> const struct audioformat *sync_fmt; >> @@ -499,7 +506,7 @@ static int snd_usb_hw_params(struct >> snd_pcm_substream *substream, >> if (fmt->implicit_fb) { >> sync_fmt = snd_usb_find_implicit_fb_sync_format(chip, fmt, >> hw_params, >> - !substream->stream, >> + !subs->direction, >> &sync_fixed_rate); >> if (!sync_fmt) { >> usb_audio_dbg(chip, >> @@ -579,15 +586,28 @@ static int snd_usb_hw_params(struct >> snd_pcm_substream *substream, >> return ret; >> } >> +EXPORT_SYMBOL_GPL(snd_usb_attach_endpoints); >> /* >> - * hw_free callback >> + * hw_params callback >> * >> - * reset the audio format and release the buffer >> + * allocate a buffer and set the given audio format. >> + * >> + * so far we use a physically linear buffer although packetize transfer >> + * doesn't need a continuous area. >> + * if sg buffer is supported on the later version of alsa, we'll follow >> + * that. >> */ >> -static int snd_usb_hw_free(struct snd_pcm_substream *substream) >> +static int snd_usb_hw_params(struct snd_pcm_substream *substream, >> + struct snd_pcm_hw_params *hw_params) >> { >> struct snd_usb_substream *subs = substream->runtime->private_data; >> + >> + return snd_usb_attach_endpoints(subs, hw_params); >> +} >> + >> +int snd_usb_detach_endpoint(struct snd_usb_substream *subs) >> +{ >> struct snd_usb_audio *chip = subs->stream->chip; >> snd_media_stop_pipeline(subs); >> @@ -603,6 +623,19 @@ static int snd_usb_hw_free(struct >> snd_pcm_substream *substream) >> return 0; >> } >> +EXPORT_SYMBOL_GPL(snd_usb_detach_endpoint); >> + >> +/* >> + * hw_free callback >> + * >> + * reset the audio format and release the buffer >> + */ >> +static int snd_usb_hw_free(struct snd_pcm_substream *substream) >> +{ >> + struct snd_usb_substream *subs = substream->runtime->private_data; >> + >> + return snd_usb_detach_endpoint(subs); >> +} >> /* free-wheeling mode? (e.g. dmix) */ >> static int in_free_wheeling_mode(struct snd_pcm_runtime *runtime) >> diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h >> index 388fe2ba346d..e36df3611a05 100644 >> --- a/sound/usb/pcm.h >> +++ b/sound/usb/pcm.h >> @@ -15,4 +15,15 @@ void snd_usb_preallocate_buffer(struct >> snd_usb_substream *subs); >> int snd_usb_audioformat_set_sync_ep(struct snd_usb_audio *chip, >> struct audioformat *fmt); >> +const struct audioformat * >> +snd_usb_find_format(struct list_head *fmt_list_head, snd_pcm_format_t >> format, >> + unsigned int rate, unsigned int channels, bool strict_match, >> + struct snd_usb_substream *subs); >> +const struct audioformat * >> +snd_usb_find_substream_format(struct snd_usb_substream *subs, >> + const struct snd_pcm_hw_params *params); >> + >> +int snd_usb_attach_endpoints(struct snd_usb_substream *subs, >> + struct snd_pcm_hw_params *hw_params); >> +int snd_usb_detach_endpoint(struct snd_usb_substream *subs); >> #endif /* __USBAUDIO_PCM_H */ > > Why is it multiple "endpoints" when attaching, but only one "endpoint" > when detaching? Both seem to be getting similar arguments. This should be detach endpoints, since it closes both the data ep as well as the sync ep if present. Will fix this. Thanks Wesley Cheng
On Tue, 29 Aug 2023 23:06:36 +0200, Wesley Cheng wrote: > > Allow for different platforms to be notified on USB SND connect/disconnect > seqeunces. This allows for platform USB SND modules to properly initialize > and populate internal structures with references to the USB SND chip > device. > > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > sound/usb/card.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ > sound/usb/card.h | 9 +++++++++ > 2 files changed, 54 insertions(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 1b2edc0fd2e9..067a1e82f4bf 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -118,6 +118,34 @@ MODULE_PARM_DESC(skip_validation, "Skip unit descriptor validation (default: no) > static DEFINE_MUTEX(register_mutex); > static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; > static struct usb_driver usb_audio_driver; > +static struct snd_usb_platform_ops *platform_ops; > + > +int snd_usb_register_platform_ops(struct snd_usb_platform_ops *ops) > +{ > + int ret; > + > + mutex_lock(®ister_mutex); > + if (platform_ops) { > + ret = -EEXIST; > + goto out; > + } > + > + platform_ops = ops; > +out: > + mutex_unlock(®ister_mutex); > + return 0; > +} > +EXPORT_SYMBOL_GPL(snd_usb_register_platform_ops); For adding this kind of API, please give the proper comment. Especially this API is special and need a caution, to mention that it can be used only for a single instance. Also, it should be mentioned that all callbacks are exclusive under the global register_mutex. > @@ -910,7 +938,11 @@ static int usb_audio_probe(struct usb_interface *intf, > chip->num_interfaces++; > usb_set_intfdata(intf, chip); > atomic_dec(&chip->active); > + > + if (platform_ops && platform_ops->connect_cb) > + platform_ops->connect_cb(chip); > mutex_unlock(®ister_mutex); One uncertain thing is the argument for connect_cb and disconnect_cb. Those take snd_usb_audio object, but the callback gets called per interface at each probe and disconnect. How does the callee handle multiple calls? Last but not least, the patch subject should be with "ALSA:" prefix, and in this case, at best "ALSA: usb-audio: xxx". thanks, Takashi
On Tue, 29 Aug 2023 23:06:37 +0200, Wesley Cheng wrote: > -/* > - * hw_params callback > - * > - * allocate a buffer and set the given audio format. > - * > - * so far we use a physically linear buffer although packetize transfer > - * doesn't need a continuous area. > - * if sg buffer is supported on the later version of alsa, we'll follow > - * that. > - */ > -static int snd_usb_hw_params(struct snd_pcm_substream *substream, > - struct snd_pcm_hw_params *hw_params) > +int snd_usb_attach_endpoints(struct snd_usb_substream *subs, > + struct snd_pcm_hw_params *hw_params) This doesn't only "attach" endpoints, but it does more other things that are needed for PCM hw_params procedure. I'd rather keep hw_params in the function name instead of creating completely different one. Ditto for hw_free. thanks, Takashi
On Tue, 29 Aug 2023 23:06:43 +0200, Wesley Cheng wrote: > > Several Qualcomm SoCs have a dedicated audio DSP, which has the ability to > support USB sound devices. This vendor driver will implement the required > handshaking with the DSP, in order to pass along required resources that > will be utilized by the DSP's USB SW. The communication channel used for > this handshaking will be using the QMI protocol. Required resources > include: > - Allocated secondary event ring address > - EP transfer ring address > - Interrupter number > > The above information will allow for the audio DSP to execute USB transfers > over the USB bus. It will also be able to support devices that have an > implicit feedback and sync endpoint as well. Offloading these data > transfers will allow the main/applications processor to enter lower CPU > power modes, and sustain a longer duration in those modes. > > Audio offloading is initiated with the following sequence: > 1. Userspace configures to route audio playback to USB backend and starts > playback on the platform soundcard. > 2. The Q6DSP AFE will communicate to the audio DSP to start the USB AFE > port. > 3. This results in a QMI packet with a STREAM enable command. > 4. The QC audio offload driver will fetch the required resources, and pass > this information as part of the QMI response to the STREAM enable command. > 5. Once the QMI response is received the audio DSP will start queuing data > on the USB bus. > > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > sound/usb/Kconfig | 15 + > sound/usb/Makefile | 2 +- > sound/usb/qcom/Makefile | 2 + > sound/usb/qcom/qc_audio_offload.c | 1813 +++++++++++++++++++++++++++++ > 4 files changed, 1831 insertions(+), 1 deletion(-) > create mode 100644 sound/usb/qcom/Makefile > create mode 100644 sound/usb/qcom/qc_audio_offload.c > > diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig > index 4a9569a3a39a..da5838656baa 100644 > --- a/sound/usb/Kconfig > +++ b/sound/usb/Kconfig > @@ -176,6 +176,21 @@ config SND_BCD2000 > To compile this driver as a module, choose M here: the module > will be called snd-bcd2000. > > +config QC_USB_AUDIO_OFFLOAD Keep SND_ prefix for consistency. And, at best, align with the module name. > + tristate "Qualcomm Audio Offload driver" > + depends on QCOM_QMI_HELPERS && SND_USB_AUDIO && USB_XHCI_SIDEBAND > + select SND_PCM > + help > + Say Y here to enable the Qualcomm USB audio offloading feature. > + > + This module sets up the required QMI stream enable/disable > + responses to requests generated by the audio DSP. It passes the > + USB transfer resource references, so that the audio DSP can issue > + USB transfers to the host controller. > + > + To compile this driver as a module, choose M here: the module > + will be called qc-audio-offload. Hmm, you renamed it differently, no? In the below: > --- /dev/null > +++ b/sound/usb/qcom/Makefile > @@ -0,0 +1,2 @@ > +snd-usb-audio-qmi-objs := usb_audio_qmi_v01.o qc_audio_offload.o > +obj-$(CONFIG_QC_USB_AUDIO_OFFLOAD) += snd-usb-audio-qmi.o ... it's called snd-usb-audio-qmi. thanks, Takashi
Hi Takashi, On 9/7/2023 8:36 AM, Takashi Iwai wrote: > On Tue, 29 Aug 2023 23:06:36 +0200, > Wesley Cheng wrote: >> >> Allow for different platforms to be notified on USB SND connect/disconnect >> seqeunces. This allows for platform USB SND modules to properly initialize >> and populate internal structures with references to the USB SND chip >> device. >> >> Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> >> --- >> sound/usb/card.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ >> sound/usb/card.h | 9 +++++++++ >> 2 files changed, 54 insertions(+) >> >> diff --git a/sound/usb/card.c b/sound/usb/card.c >> index 1b2edc0fd2e9..067a1e82f4bf 100644 >> --- a/sound/usb/card.c >> +++ b/sound/usb/card.c >> @@ -118,6 +118,34 @@ MODULE_PARM_DESC(skip_validation, "Skip unit descriptor validation (default: no) >> static DEFINE_MUTEX(register_mutex); >> static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; >> static struct usb_driver usb_audio_driver; >> +static struct snd_usb_platform_ops *platform_ops; >> + >> +int snd_usb_register_platform_ops(struct snd_usb_platform_ops *ops) >> +{ >> + int ret; >> + >> + mutex_lock(®ister_mutex); >> + if (platform_ops) { >> + ret = -EEXIST; >> + goto out; >> + } >> + >> + platform_ops = ops; >> +out: >> + mutex_unlock(®ister_mutex); >> + return 0; >> +} >> +EXPORT_SYMBOL_GPL(snd_usb_register_platform_ops); > > For adding this kind of API, please give the proper comment. > Especially this API is special and need a caution, to mention that it > can be used only for a single instance. > > Also, it should be mentioned that all callbacks are exclusive under > the global register_mutex. > Thanks for taking the time to review. Sure, I'll add some comments in these new APIs to document what they are used for and how they are protected and limited. >> @@ -910,7 +938,11 @@ static int usb_audio_probe(struct usb_interface *intf, >> chip->num_interfaces++; >> usb_set_intfdata(intf, chip); >> atomic_dec(&chip->active); >> + >> + if (platform_ops && platform_ops->connect_cb) >> + platform_ops->connect_cb(chip); >> mutex_unlock(®ister_mutex); > > One uncertain thing is the argument for connect_cb and disconnect_cb. > Those take snd_usb_audio object, but the callback gets called per > interface at each probe and disconnect. How does the callee handle > multiple calls? I guess it should depend on how the platform driver wants to handle it? I haven't run into a device with multiple UAC interfaces before, so I'll need to mimic this configuration on a device, so I can see how it exposes itself. Will investigate this a bit more on my end and come back with my findings. > > Last but not least, the patch subject should be with "ALSA:" prefix, > and in this case, at best "ALSA: usb-audio: xxx". > > Got it, thanks! Thanks Wesley Cheng
Hi Takashi, On 9/7/2023 8:38 AM, Takashi Iwai wrote: > On Tue, 29 Aug 2023 23:06:37 +0200, > Wesley Cheng wrote: >> -/* >> - * hw_params callback >> - * >> - * allocate a buffer and set the given audio format. >> - * >> - * so far we use a physically linear buffer although packetize transfer >> - * doesn't need a continuous area. >> - * if sg buffer is supported on the later version of alsa, we'll follow >> - * that. >> - */ >> -static int snd_usb_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *hw_params) >> +int snd_usb_attach_endpoints(struct snd_usb_substream *subs, >> + struct snd_pcm_hw_params *hw_params) > > This doesn't only "attach" endpoints, but it does more other things > that are needed for PCM hw_params procedure. I'd rather keep > hw_params in the function name instead of creating completely > different one. > > Ditto for hw_free. > Sure I'll keep the same nomenclature as it was previously. Thanks Wesley Cheng
Hi Takashi, On 9/7/2023 8:51 AM, Takashi Iwai wrote: > On Tue, 29 Aug 2023 23:06:43 +0200, > Wesley Cheng wrote: >> >> Several Qualcomm SoCs have a dedicated audio DSP, which has the ability to >> support USB sound devices. This vendor driver will implement the required >> handshaking with the DSP, in order to pass along required resources that >> will be utilized by the DSP's USB SW. The communication channel used for >> this handshaking will be using the QMI protocol. Required resources >> include: >> - Allocated secondary event ring address >> - EP transfer ring address >> - Interrupter number >> >> The above information will allow for the audio DSP to execute USB transfers >> over the USB bus. It will also be able to support devices that have an >> implicit feedback and sync endpoint as well. Offloading these data >> transfers will allow the main/applications processor to enter lower CPU >> power modes, and sustain a longer duration in those modes. >> >> Audio offloading is initiated with the following sequence: >> 1. Userspace configures to route audio playback to USB backend and starts >> playback on the platform soundcard. >> 2. The Q6DSP AFE will communicate to the audio DSP to start the USB AFE >> port. >> 3. This results in a QMI packet with a STREAM enable command. >> 4. The QC audio offload driver will fetch the required resources, and pass >> this information as part of the QMI response to the STREAM enable command. >> 5. Once the QMI response is received the audio DSP will start queuing data >> on the USB bus. >> >> Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> >> --- >> sound/usb/Kconfig | 15 + >> sound/usb/Makefile | 2 +- >> sound/usb/qcom/Makefile | 2 + >> sound/usb/qcom/qc_audio_offload.c | 1813 +++++++++++++++++++++++++++++ >> 4 files changed, 1831 insertions(+), 1 deletion(-) >> create mode 100644 sound/usb/qcom/Makefile >> create mode 100644 sound/usb/qcom/qc_audio_offload.c >> >> diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig >> index 4a9569a3a39a..da5838656baa 100644 >> --- a/sound/usb/Kconfig >> +++ b/sound/usb/Kconfig >> @@ -176,6 +176,21 @@ config SND_BCD2000 >> To compile this driver as a module, choose M here: the module >> will be called snd-bcd2000. >> >> +config QC_USB_AUDIO_OFFLOAD > > Keep SND_ prefix for consistency. And, at best, align with the module > name. > >> + tristate "Qualcomm Audio Offload driver" >> + depends on QCOM_QMI_HELPERS && SND_USB_AUDIO && USB_XHCI_SIDEBAND >> + select SND_PCM >> + help >> + Say Y here to enable the Qualcomm USB audio offloading feature. >> + >> + This module sets up the required QMI stream enable/disable >> + responses to requests generated by the audio DSP. It passes the >> + USB transfer resource references, so that the audio DSP can issue >> + USB transfers to the host controller. >> + >> + To compile this driver as a module, choose M here: the module >> + will be called qc-audio-offload. > > Hmm, you renamed it differently, no? In the below: > >> --- /dev/null >> +++ b/sound/usb/qcom/Makefile >> @@ -0,0 +1,2 @@ >> +snd-usb-audio-qmi-objs := usb_audio_qmi_v01.o qc_audio_offload.o >> +obj-$(CONFIG_QC_USB_AUDIO_OFFLOAD) += snd-usb-audio-qmi.o > > ... it's called snd-usb-audio-qmi. > Will fix this, thanks. Thanks Wesley Cheng
Hi Takashi, On 9/11/2023 10:57 AM, Wesley Cheng wrote: > Hi Takashi, > > On 9/7/2023 8:36 AM, Takashi Iwai wrote: >> On Tue, 29 Aug 2023 23:06:36 +0200, >> Wesley Cheng wrote: >>> >>> Allow for different platforms to be notified on USB SND >>> connect/disconnect >>> seqeunces. This allows for platform USB SND modules to properly >>> initialize >>> and populate internal structures with references to the USB SND chip >>> device. >>> >>> Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> >>> --- >>> sound/usb/card.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ >>> sound/usb/card.h | 9 +++++++++ >>> 2 files changed, 54 insertions(+) >>> >>> diff --git a/sound/usb/card.c b/sound/usb/card.c >>> index 1b2edc0fd2e9..067a1e82f4bf 100644 >>> --- a/sound/usb/card.c >>> +++ b/sound/usb/card.c >>> @@ -118,6 +118,34 @@ MODULE_PARM_DESC(skip_validation, "Skip unit >>> descriptor validation (default: no) >>> static DEFINE_MUTEX(register_mutex); >>> static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; >>> static struct usb_driver usb_audio_driver; >>> +static struct snd_usb_platform_ops *platform_ops; >>> + >>> +int snd_usb_register_platform_ops(struct snd_usb_platform_ops *ops) >>> +{ >>> + int ret; >>> + >>> + mutex_lock(®ister_mutex); >>> + if (platform_ops) { >>> + ret = -EEXIST; >>> + goto out; >>> + } >>> + >>> + platform_ops = ops; >>> +out: >>> + mutex_unlock(®ister_mutex); >>> + return 0; >>> +} >>> +EXPORT_SYMBOL_GPL(snd_usb_register_platform_ops); >> >> For adding this kind of API, please give the proper comment. >> Especially this API is special and need a caution, to mention that it >> can be used only for a single instance. >> >> Also, it should be mentioned that all callbacks are exclusive under >> the global register_mutex. >> > > Thanks for taking the time to review. Sure, I'll add some comments in > these new APIs to document what they are used for and how they are > protected and limited. > >>> @@ -910,7 +938,11 @@ static int usb_audio_probe(struct usb_interface >>> *intf, >>> chip->num_interfaces++; >>> usb_set_intfdata(intf, chip); >>> atomic_dec(&chip->active); >>> + >>> + if (platform_ops && platform_ops->connect_cb) >>> + platform_ops->connect_cb(chip); >>> mutex_unlock(®ister_mutex); >> >> One uncertain thing is the argument for connect_cb and disconnect_cb. >> Those take snd_usb_audio object, but the callback gets called per >> interface at each probe and disconnect. How does the callee handle >> multiple calls? > > I guess it should depend on how the platform driver wants to handle it? > I haven't run into a device with multiple UAC interfaces before, so > I'll need to mimic this configuration on a device, so I can see how it > exposes itself. > > Will investigate this a bit more on my end and come back with my findings. > So looks like if there is a device that has multiple UAC interfaces, then it just results in one USB sound card with multiple USB streams. As of now, we do expose a ksndcontrol that allows for userspace to specify which card and PCM device to issue the audio playback on. Then based on the audio profile received by the audio DSP, it will narrow it down to see if that substream supports the requested profile. This selection is done in the DPCM backend driver (q6usb), since the overall QMI USB stream request stems from the audio dsp, and will carry this information about which PCM device and card the USB offload driver should use. This may differ based on the implementation of the offload path, hence why I mentioned that it might be platform specific. There is one improvement I need to make in our QC offload driver to accommodate for this scenario, and that is to avoid having to re-register for the XHCI sideband if there is already a USB SND card allocated, which I will fix and submit in the next revision. Thanks Wesley Cheng
Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feature_interrupters - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feature_interrupters Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (3): xhci: add support to allocate several interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (29): usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: Export USB SND APIs for modules dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition dt-bindings: usb: xhci: Add num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present sound: usb: qcom: Add USB QMI definitions sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: q6afe: Split USB AFE dev_token param into separate API sound: Pass USB SND card and PCM information to SOC USB sound: soc: qdsp6: Add SND kcontrol to select offload device sound: soc: qdsp6: Add SND kcontrol for fetching offload status sound: soc: qcom: q6usb: Add headphone jack for offload connection status sound: usb: qc_audio_offload: Use card and PCM index from QMI request sound: usb: card: Allow for rediscovery of connected USB SND devices sound: soc: soc-usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,q6usb-dais.yaml | 53 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 13 + .../devicetree/bindings/usb/usb-xhci.yaml | 4 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 4 + drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-hub.c | 29 +- drivers/usb/host/xhci-mem.c | 102 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 48 +- drivers/usb/host/xhci-sideband.c | 195 ++ drivers/usb/host/xhci.c | 111 +- drivers/usb/host/xhci.h | 105 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 86 + include/linux/usb/xhci-sideband.h | 57 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 51 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 55 + sound/soc/qcom/qdsp6/q6afe.c | 206 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 452 ++++ sound/soc/soc-usb.c | 200 ++ sound/usb/Kconfig | 15 + sound/usb/Makefile | 2 +- sound/usb/card.c | 108 + sound/usb/card.h | 23 + sound/usb/endpoint.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 86 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1837 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 45 files changed, 4863 insertions(+), 192 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-intr.h create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h