Re: [PATCH v5 12/32] sound: usb: Export USB SND APIs for modules

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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@xxxxxxxxxxx>
---
  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.



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