audioaggregator: Only check downstream caps when handling CAPS events if we didn...

If we already negotiated with downstream there is not point in checking
if the caps are supported. We already know that this is the case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/768>

diff --git a/gst-libs/gst/audio/gstaudioaggregator.c b/gst-libs/gst/audio/gstaudioaggregator.c
index 546c9a0..5bc0cbd 100644 (file)
--- a/gst-libs/gst/audio/gstaudioaggregator.c
+++ b/gst-libs/gst/audio/gstaudioaggregator.c
@@ -869,20 +869,9 @@ gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
 {
   GstAudioAggregatorPad *first_configured_pad =
       gst_audio_aggregator_get_first_configured_pad (agg);
-  GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
-  GstCaps *rate_caps;
   GstAudioInfo info;
   gboolean ret = TRUE;
-  gboolean downstream_supports_rate;
-
-  /* Returns NULL if there is no downstream peer */
-  if (!downstream_caps)
-    downstream_caps = gst_pad_get_pad_template_caps (agg->srcpad);
-
-  if (gst_caps_is_empty (downstream_caps)) {
-    ret = FALSE;
-    goto done;
-  }
+  gboolean downstream_supports_rate = TRUE;
 
   if (!gst_audio_info_from_caps (&info, caps)) {
     GST_WARNING_OBJECT (agg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
@@ -893,11 +882,28 @@ gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
    * because offsets will have to be updated, and audio resampling
    * has a latency to take into account
    */
-  rate_caps =
-      gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, info.rate, NULL);
-  downstream_supports_rate =
-      gst_caps_can_intersect (rate_caps, downstream_caps);
-  gst_caps_unref (rate_caps);
+
+  /* Only check against the downstream caps if we didn't configure any caps
+   * so far. Otherwise we already know that downstream supports the rate
+   * because we negotiated with downstream */
+  if (!first_configured_pad) {
+    GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
+
+    /* Returns NULL if there is no downstream peer */
+    if (downstream_caps) {
+      GstCaps *rate_caps =
+          gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, info.rate,
+          NULL);
+
+      gst_caps_set_features_simple (rate_caps,
+          gst_caps_features_copy (GST_CAPS_FEATURES_ANY));
+
+      downstream_supports_rate =
+          gst_caps_can_intersect (rate_caps, downstream_caps);
+      gst_caps_unref (rate_caps);
+      gst_caps_unref (downstream_caps);
+    }
+  }
 
   if (!downstream_supports_rate || (first_configured_pad
           && info.rate != first_configured_pad->info.rate)) {
@@ -913,13 +919,9 @@ gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
     GST_OBJECT_UNLOCK (aaggpad);
   }
 
-done:
   if (first_configured_pad)
     gst_object_unref (first_configured_pad);
 
-  if (downstream_caps)
-    gst_caps_unref (downstream_caps);
-
   return ret;
 }