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: > 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.
On Tue, 29 Aug 2023 14:06:49 -0700, Wesley Cheng wrote: > Add an example on enabling of USB offload for the Q6DSP. The routing can > be done by the mixer, which can pass the multimedia stream to the USB > backend. > > Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com> > --- > .../devicetree/bindings/sound/qcom,sm8250.yaml | 15 +++++++++++++++ > 1 file changed, 15 insertions(+) > Acked-by: Rob Herring <robh@kernel.org>