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Message-ID: <20250401075323.GH98772@ubuntu>
Date: Tue, 1 Apr 2025 16:53:23 +0900
From: Jung Daehwan <dh10.jung@...sung.com>
To: Puma Hsu <pumahsu@...gle.com>
Cc: Wesley Cheng <quic_wcheng@...cinc.com>, srinivas.kandagatla@...aro.org,
mathias.nyman@...el.com, perex@...ex.cz, conor+dt@...nel.org,
dmitry.torokhov@...il.com, corbet@....net, broonie@...nel.org,
lgirdwood@...il.com, krzk+dt@...nel.org,
pierre-louis.bossart@...ux.intel.com, Thinh.Nguyen@...opsys.com,
tiwai@...e.com, robh@...nel.org, gregkh@...uxfoundation.org,
linux-kernel@...r.kernel.org, devicetree@...r.kernel.org,
linux-sound@...r.kernel.org, linux-input@...r.kernel.org,
linux-usb@...r.kernel.org, linux-arm-msm@...r.kernel.org,
linux-doc@...r.kernel.org
Subject: Re: [PATCH v36 08/31] ALSA: usb-audio: Export USB SND APIs for
modules
On Fri, Mar 28, 2025 at 03:47:00PM +0800, Puma Hsu wrote:
> On Wed, Mar 19, 2025 at 8:52 AM Wesley Cheng <quic_wcheng@...cinc.com> wrote:
> >
> > Some vendor modules will utilize useful parsing and endpoint management
> > APIs to start audio playback/capture.
> >
> > Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@...ux.intel.com>
> > Signed-off-by: Wesley Cheng <quic_wcheng@...cinc.com>
>
> Tested-by: Puma Hsu <pumahsu@...gle.com>
>
Tested-by: Daehwan Jung <dh10.jung@...sung.com>
> > ---
> > sound/usb/card.c | 4 +++
> > sound/usb/endpoint.c | 1 +
> > sound/usb/helper.c | 1 +
> > sound/usb/pcm.c | 75 +++++++++++++++++++++++++++++++-------------
> > sound/usb/pcm.h | 11 +++++++
> > 5 files changed, 71 insertions(+), 21 deletions(-)
> >
> > diff --git a/sound/usb/card.c b/sound/usb/card.c
> > index 9c411b82a218..a0087bde684c 100644
> > --- a/sound/usb/card.c
> > +++ b/sound/usb/card.c
> > @@ -1030,6 +1030,7 @@ 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)
> > @@ -1038,6 +1039,7 @@ 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)
> > {
> > @@ -1060,6 +1062,7 @@ 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)
> > {
> > @@ -1073,6 +1076,7 @@ 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 a29f28eb7d0c..1fed039b10ed 100644
> > --- a/sound/usb/endpoint.c
> > +++ b/sound/usb/endpoint.c
> > @@ -1524,6 +1524,7 @@ 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 72b671fb2c84..497d2b27fb59 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..18467da6fd9e 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_hw_params(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_hw_params);
> >
> > /*
> > - * 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_pcm_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_hw_params(subs, hw_params);
> > +}
> > +
> > +int snd_usb_hw_free(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_hw_free);
> > +
> > +/*
> > + * hw_free callback
> > + *
> > + * reset the audio format and release the buffer
> > + */
> > +static int snd_usb_pcm_hw_free(struct snd_pcm_substream *substream)
> > +{
> > + struct snd_usb_substream *subs = substream->runtime->private_data;
> > +
> > + return snd_usb_hw_free(subs);
> > +}
> >
> > /* free-wheeling mode? (e.g. dmix) */
> > static int in_free_wheeling_mode(struct snd_pcm_runtime *runtime)
> > @@ -1746,8 +1779,8 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
> > static const struct snd_pcm_ops snd_usb_playback_ops = {
> > .open = snd_usb_pcm_open,
> > .close = snd_usb_pcm_close,
> > - .hw_params = snd_usb_hw_params,
> > - .hw_free = snd_usb_hw_free,
> > + .hw_params = snd_usb_pcm_hw_params,
> > + .hw_free = snd_usb_pcm_hw_free,
> > .prepare = snd_usb_pcm_prepare,
> > .trigger = snd_usb_substream_playback_trigger,
> > .sync_stop = snd_usb_pcm_sync_stop,
> > @@ -1758,8 +1791,8 @@ static const struct snd_pcm_ops snd_usb_playback_ops = {
> > static const struct snd_pcm_ops snd_usb_capture_ops = {
> > .open = snd_usb_pcm_open,
> > .close = snd_usb_pcm_close,
> > - .hw_params = snd_usb_hw_params,
> > - .hw_free = snd_usb_hw_free,
> > + .hw_params = snd_usb_pcm_hw_params,
> > + .hw_free = snd_usb_pcm_hw_free,
> > .prepare = snd_usb_pcm_prepare,
> > .trigger = snd_usb_substream_capture_trigger,
> > .sync_stop = snd_usb_pcm_sync_stop,
> > diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h
> > index 388fe2ba346d..c096021adb2b 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_hw_params(struct snd_usb_substream *subs,
> > + struct snd_pcm_hw_params *hw_params);
> > +int snd_usb_hw_free(struct snd_usb_substream *subs);
> > #endif /* __USBAUDIO_PCM_H */
> >
>
>
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