webrtc: fix rtx + bundle

If bundle was used in combination with rtx, only the bundled transport
stream would have correctly configured rtx parameters.

Iterate over the payloads upfront in the bundled case to ensure the
correct payload mapping is set for the RTX elements.

1 files changed, 85 insertions, 71 deletions

diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index 472f779a5..29294cfcc 100644
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -2962,6 +2962,75 @@ _filter_sdp_fields (GQuark field_id, const GValue * value,
 }
 
 static void
+_update_transport_ptmap_from_media (GstWebRTCBin * webrtc,
+    TransportStream * stream, const GstSDPMessage * sdp, guint media_idx)
+{
+  guint i, len;
+  const gchar *proto;
+  const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
+
+  /* get proto */
+  proto = gst_sdp_media_get_proto (media);
+  if (proto != NULL) {
+    /* Parse global SDP attributes once */
+    GstCaps *global_caps = gst_caps_new_empty_simple ("application/x-unknown");
+    GST_DEBUG_OBJECT (webrtc, "mapping sdp session level attributes to caps");
+    gst_sdp_message_attributes_to_caps (sdp, global_caps);
+    GST_DEBUG_OBJECT (webrtc, "mapping sdp media level attributes to caps");
+    gst_sdp_media_attributes_to_caps (media, global_caps);
+
+    len = gst_sdp_media_formats_len (media);
+    for (i = 0; i < len; i++) {
+      GstCaps *caps, *outcaps;
+      GstStructure *s;
+      PtMapItem item;
+      gint pt;
+      guint j;
+
+      pt = atoi (gst_sdp_media_get_format (media, i));
+
+      GST_DEBUG_OBJECT (webrtc, " looking at %d pt: %d", i, pt);
+
+      /* convert caps */
+      caps = gst_sdp_media_get_caps_from_media (media, pt);
+      if (caps == NULL) {
+        GST_WARNING_OBJECT (webrtc, " skipping pt %d without caps", pt);
+        continue;
+      }
+
+      /* Merge in global caps */
+      /* Intersect will merge in missing fields to the current caps */
+      outcaps = gst_caps_intersect (caps, global_caps);
+      gst_caps_unref (caps);
+
+      s = gst_caps_get_structure (outcaps, 0);
+      gst_structure_set_name (s, "application/x-rtp");
+      if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
+        gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
+
+      item.caps = gst_caps_new_empty ();
+
+      for (j = 0; j < gst_caps_get_size (outcaps); j++) {
+        GstStructure *s = gst_caps_get_structure (outcaps, j);
+        GstStructure *filtered =
+            gst_structure_new_empty (gst_structure_get_name (s));
+
+        gst_structure_foreach (s,
+            (GstStructureForeachFunc) _filter_sdp_fields, filtered);
+        gst_caps_append_structure (item.caps, filtered);
+      }
+
+      item.pt = pt;
+      gst_caps_unref (outcaps);
+
+      g_array_append_val (stream->ptmap, item);
+    }
+
+    gst_caps_unref (global_caps);
+  }
+}
+
+static void
 _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
     const GstSDPMessage * sdp, guint media_idx,
     TransportStream * stream, GstWebRTCRTPTransceiver * rtp_trans,
@@ -2990,9 +3059,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
     const GstSDPMedia *local_media, *remote_media;
     GstWebRTCRTPTransceiverDirection local_dir, remote_dir;
     GstWebRTCDTLSSetup local_setup, remote_setup;
-    guint i, len;
-    const gchar *proto;
-    GstCaps *global_caps;
 
     local_media =
         gst_sdp_message_get_media (webrtc->current_local_description->sdp,
@@ -3014,72 +3080,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
     if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE)
       return;
 
-    /* get proto */
-    proto = gst_sdp_media_get_proto (media);
-    if (proto != NULL) {
-      /* Parse global SDP attributes once */
-      global_caps = gst_caps_new_empty_simple ("application/x-unknown");
-      GST_DEBUG_OBJECT (webrtc, "mapping sdp session level attributes to caps");
-      gst_sdp_message_attributes_to_caps (sdp, global_caps);
-      GST_DEBUG_OBJECT (webrtc, "mapping sdp media level attributes to caps");
-      gst_sdp_media_attributes_to_caps (media, global_caps);
-
-      if (!bundled) {
-        /* clear the ptmap */
-        g_array_set_size (stream->ptmap, 0);
-      }
-
-      len = gst_sdp_media_formats_len (media);
-      for (i = 0; i < len; i++) {
-        GstCaps *caps, *outcaps;
-        GstStructure *s;
-        PtMapItem item;
-        gint pt;
-        guint j;
-
-        pt = atoi (gst_sdp_media_get_format (media, i));
-
-        GST_DEBUG_OBJECT (webrtc, " looking at %d pt: %d", i, pt);
-
-        /* convert caps */
-        caps = gst_sdp_media_get_caps_from_media (media, pt);
-        if (caps == NULL) {
-          GST_WARNING_OBJECT (webrtc, " skipping pt %d without caps", pt);
-          continue;
-        }
-
-        /* Merge in global caps */
-        /* Intersect will merge in missing fields to the current caps */
-        outcaps = gst_caps_intersect (caps, global_caps);
-        gst_caps_unref (caps);
-
-        s = gst_caps_get_structure (outcaps, 0);
-        gst_structure_set_name (s, "application/x-rtp");
-        if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
-                "ULPFEC"))
-          gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
-
-        item.caps = gst_caps_new_empty ();
-
-        for (j = 0; j < gst_caps_get_size (outcaps); j++) {
-          GstStructure *s = gst_caps_get_structure (outcaps, j);
-          GstStructure *filtered =
-              gst_structure_new_empty (gst_structure_get_name (s));
-
-          gst_structure_foreach (s,
-              (GstStructureForeachFunc) _filter_sdp_fields, filtered);
-          gst_caps_append_structure (item.caps, filtered);
-        }
-
-        item.pt = pt;
-        gst_caps_unref (outcaps);
-
-        g_array_append_val (stream->ptmap, item);
-      }
-
-      gst_caps_unref (global_caps);
-    }
-
     if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
         && prev_dir != new_dir) {
       GST_FIXME_OBJECT (webrtc, "implement transceiver direction changes");
@@ -3381,9 +3381,12 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
     bundle_stream = _get_or_create_transport_stream (webrtc, bundle_idx,
         _message_media_is_datachannel (sdp->sdp, bundle_idx));
 
-    g_array_set_size (bundle_stream->ptmap, 0);
-
     _connect_rtpfunnel (webrtc, bundle_idx);
+
+    g_array_set_size (bundle_stream->ptmap, 0);
+    for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
+      _update_transport_ptmap_from_media (webrtc, bundle_stream, sdp->sdp, i);
+    }
   }
 
   for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
@@ -3405,6 +3408,10 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
 
     stream = _get_or_create_transport_stream (webrtc, transport_idx,
         _message_media_is_datachannel (sdp->sdp, transport_idx));
+    if (!bundled) {
+      g_array_set_size (stream->ptmap, 0);
+      _update_transport_ptmap_from_media (webrtc, stream, sdp->sdp, i);
+    }
 
     if (trans)
       webrtc_transceiver_set_transport ((WebRTCTransceiver *) trans, stream);
@@ -4277,6 +4284,10 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
     }
   }
 
+  GST_LOG_OBJECT (webrtc, "requesting aux sender for stream %" GST_PTR_FORMAT
+      " with transport %" GST_PTR_FORMAT " and pt map %" GST_PTR_FORMAT, stream,
+      trans, pt_map);
+
   if (gst_structure_n_fields (pt_map)) {
     GstElement *rtx;
     GstPad *pad;
@@ -4330,6 +4341,9 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
     rtx_pt = transport_stream_get_pt (stream, "RTX");
   }
 
+  GST_LOG_OBJECT (webrtc, "requesting aux receiver for stream %" GST_PTR_FORMAT
+      " with pt red:%u rtx:%u", stream, red_pt, rtx_pt);
+
   if (red_pt || rtx_pt)
     ret = gst_bin_new (NULL);