audioaggregator: Sync property values to output timestamp

This is what videoaggregator already does since 2019, and it makes
sense. The properties need to change at every output frame based on
the output time because they may change even though the input frame is
not changing. See:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/commit/6a8c15f3bdc7077b1e5954580c25f99e1b112ead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3851>

gst-libs/gst/audio/gstaudioaggregator.c

@@ -521,7 +521,6 @@ static GstBuffer *gst_audio_aggregator_do_clip (GstAggregator * agg,
     GstAggregatorPad * bpad, GstBuffer * buffer);
 static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg,
     gboolean timeout);
-static gboolean sync_pad_values (GstElement * aagg, GstPad * pad, gpointer ud);
 static gboolean gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg,
     GstCaps * caps);
 static GstFlowReturn
@@ -2150,30 +2149,6 @@ gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
   return outbuf;
 }
 
-static gboolean
-sync_pad_values (GstElement * aagg, GstPad * pad, gpointer user_data)
-{
-  GstAudioAggregatorPad *aapad = GST_AUDIO_AGGREGATOR_PAD (pad);
-  GstAggregatorPad *bpad = GST_AGGREGATOR_PAD_CAST (pad);
-  GstClockTime timestamp, stream_time;
-
-  if (aapad->priv->buffer == NULL)
-    return TRUE;
-
-  timestamp = GST_BUFFER_PTS (aapad->priv->buffer);
-  GST_OBJECT_LOCK (bpad);
-  stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
-      timestamp);
-  GST_OBJECT_UNLOCK (bpad);
-
-  /* sync object properties on stream time */
-  /* TODO: Ideally we would want to do that on every sample */
-  if (GST_CLOCK_TIME_IS_VALID (stream_time))
-    gst_object_sync_values (GST_OBJECT_CAST (pad), stream_time);
-
-  return TRUE;
-}
-
 static GstSample *
 gst_audio_aggregator_peek_next_sample (GstAggregator * agg,
     GstAggregatorPad * aggpad)
@@ -2201,6 +2176,18 @@ gst_audio_aggregator_peek_next_sample (GstAggregator * agg,
   return sample;
 }
 
+static gboolean
+sync_pad_values (GstElement * aagg, GstPad * pad, gpointer user_data)
+{
+  gint64 *outbuf_stream_time = user_data;
+
+  /* sync object properties on stream time */
+  if (GST_CLOCK_TIME_IS_VALID (*outbuf_stream_time))
+    gst_object_sync_values (GST_OBJECT_CAST (pad), *outbuf_stream_time);
+
+  return TRUE;
+}
+
 static GstFlowReturn
 gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
 {
@@ -2244,13 +2231,11 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
   guint blocksize;
   GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
   GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
+  GstClockTime outbuf_stream_time;
 
   element = GST_ELEMENT (agg);
   aagg = GST_AUDIO_AGGREGATOR (agg);
 
-  /* Sync pad properties to the stream time */
-  gst_element_foreach_sink_pad (element, sync_pad_values, NULL);
-
   GST_AUDIO_AGGREGATOR_LOCK (aagg);
   GST_OBJECT_LOCK (agg);
 
@@ -2479,8 +2464,20 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
         aagg->priv->selected_samples_info);
   }
 
-  GST_OBJECT_LOCK (agg);
+  /* Calculate the stream time of the output buffer using its PTS. See below
+   * near gst_aggregator_finish_buffer() for the calculation for that. */
+  if (agg_segment->rate > 0.0) {
+    outbuf_stream_time = gst_segment_to_stream_time (agg_segment,
+        GST_FORMAT_TIME, agg_segment->position);
+  } else {
+    outbuf_stream_time = gst_segment_to_stream_time (agg_segment,
+        GST_FORMAT_TIME, next_timestamp);
+  }
+
+  /* Sync pad properties to the stream time */
+  gst_element_foreach_sink_pad (element, sync_pad_values, &outbuf_stream_time);
 
+  GST_OBJECT_LOCK (agg);
   // mix_buffer() will shortly release the object lock so we need to
   // ensure that the pad list stays valid.
   n_sinkpads = element->numsinkpads;