webrtcdsp: Deal with echo probe info not being available

Even if we don't yet know what the echo probe format is, we want to be able to
provide silence for the reverse path, so that when the probe becomes available,
there is no ambiguity around what time period the new set of samples are for.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>

subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp

@@ -370,6 +370,8 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
   GstWebrtcEchoProbe *probe = NULL;
   webrtc::AudioProcessing *apm;
   GstBuffer *buf = NULL;
+  GstAudioInfo info;
+  gboolean interleaved = self->interleaved;
   GstAudioBuffer abuf;
   GstFlowReturn ret = GST_FLOW_OK;
   gint err, delay;
@@ -377,36 +379,38 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
   GST_OBJECT_LOCK (self);
   if (self->echo_cancel)
     probe = GST_WEBRTC_ECHO_PROBE (g_object_ref (self->probe));
+  info = self->info;
   GST_OBJECT_UNLOCK (self);
 
   /* If echo cancellation is disabled */
   if (!probe)
     return GST_FLOW_OK;
 
-  webrtc::StreamConfig config (probe->info.rate, probe->info.channels,
-      false);
-  apm = self->apm;
+  delay =
+    gst_webrtc_echo_probe_read (probe, rec_time, &buf, &info, &interleaved);
 
-  delay = gst_webrtc_echo_probe_read (probe, rec_time, &buf);
+  apm = self->apm;
   apm->set_stream_delay_ms (delay);
 
+  webrtc::StreamConfig config (info.rate, info.channels, false);
+
   g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
 
   if (delay < 0)
     goto done;
 
-  if (probe->info.rate != self->info.rate) {
+  if (info.rate != self->info.rate) {
     GST_ELEMENT_ERROR (self, STREAM, FORMAT,
         ("Echo Probe has rate %i , while the DSP is running at rate %i,"
          " use a caps filter to ensure those are the same.",
-         probe->info.rate, self->info.rate), (NULL));
+         info.rate, self->info.rate), (NULL));
     ret = GST_FLOW_ERROR;
     goto done;
   }
 
-  gst_audio_buffer_map (&abuf, &probe->info, buf, GST_MAP_READWRITE);
+  gst_audio_buffer_map (&abuf, &info, buf, GST_MAP_READWRITE);
 
-  if (probe->interleaved) {
+  if (interleaved) {
     int16_t * const data = (int16_t * const) abuf.planes[0];
 
     if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)

subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp

@@ -304,7 +304,7 @@ gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
 
 gint
 gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
-    GstBuffer ** buf)
+    GstBuffer ** buf, GstAudioInfo * info, gboolean * interleaved)
 {
   GstClockTimeDiff diff;
   gsize avail, skip, offset, size = 0;
@@ -315,10 +315,17 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
   /* We always return a buffer -- if don't have data (size == 0), we generate a
    * silence buffer */
 
-  if (!GST_CLOCK_TIME_IS_VALID (self->latency) ||
-      !GST_AUDIO_INFO_IS_VALID (&self->info))
+  if (!GST_CLOCK_TIME_IS_VALID (self->latency))
     goto copy;
 
+  /* If we have a format, use that, else generate silence in input format */
+  if (!GST_AUDIO_INFO_IS_VALID (&self->info)) {
+    goto copy;
+  } else {
+   *info = self->info;
+   *interleaved = self->interleaved;
+  }
+
   if (self->interleaved)
     avail = gst_adapter_available (self->adapter) / self->info.bpf;
   else
@@ -375,11 +382,14 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
 
 copy:
   if (!size) {
-    /* No data, provide a period's worth of silence */
-    *buf = gst_buffer_new_allocate (NULL, self->period_size, NULL);
-    gst_buffer_memset (*buf, 0, 0, self->period_size);
-    gst_buffer_add_audio_meta (*buf, &self->info, self->period_samples,
-        NULL);
+    /* No data, provide a period's worth of silence, using our format if we have
+     * it, or the provided format if we don't */
+    guint period_samples = info->rate / 100;
+    guint period_size = period_samples * info->bpf;
+
+    *buf = gst_buffer_new_allocate (NULL, period_size, NULL);
+    gst_buffer_memset (*buf, 0, 0, period_size);
+    gst_buffer_add_audio_meta (*buf, info, period_samples, NULL);
   } else {
     /* We have some actual data, pop period_samples' worth if have it, else pad
      * with silence and provide what we do have */

subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h

@@ -99,7 +99,8 @@ GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
 GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
 void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
 gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
-    GstClockTime rec_time, GstBuffer ** buf);
+    GstClockTime rec_time, GstBuffer ** buf, GstAudioInfo * info,
+    gboolean * interleaved);
 
 G_END_DECLS
 #endif /* __GST_WEBRTC_ECHO_PROBE_H__ */