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Message-ID: <CAGCq0LYs5=Re74G99n3=Vbdu1Q2M5-SKZqXTgPKJZYtSHjXZPA@mail.gmail.com>
Date: Fri, 28 Mar 2025 15:47:00 +0800
From: Puma Hsu <pumahsu@...gle.com>
To: Wesley Cheng <quic_wcheng@...cinc.com>
Cc: 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 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>
> ---
> 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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