From fcba27a9347b9610498bc410c676f5eaa09bd369 Mon Sep 17 00:00:00 2001
From: =?utf8?q?Olivier=20Cr=C3=AAte?= <olivier.crete@collabora.com>
Date: Wed, 26 Aug 2020 17:48:06 -0400
Subject: [PATCH] webrtcbin: Merge the RTX SSRCs from all transceivers when
 bundling

Change-Id: Iedc2ed9fdeecc4b0b846983957be9a4fb47c6aaa
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1545>
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---
 ext/webrtc/gstwebrtcbin.c | 37 +++++++++++++++++++++++++++----------
 1 file changed, 27 insertions(+), 10 deletions(-)

diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index c2ae10317..725b865f7 100644
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -5292,6 +5292,16 @@ unknown_session:
   }
 }
 
+static gboolean
+_merge_structure (GQuark field_id, const GValue * value, gpointer user_data)
+{
+  GstStructure *s = user_data;
+
+  gst_structure_id_set_value (s, field_id, value);
+
+  return TRUE;
+}
+
 static GstElement *
 on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
     GstWebRTCBin * webrtc)
@@ -5302,28 +5312,26 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
   GstStructure *pt_map = NULL;
 #endif
   GstElement *ret = NULL;
-  GstWebRTCRTPTransceiver *trans;
 
   stream = _find_transport_for_session (webrtc, session_id);
-  trans = _find_transceiver (webrtc, &session_id,
-      (FindTransceiverFunc) transceiver_match_for_mline);
 
   if (stream)
     have_rtx = transport_stream_get_pt (stream, "RTX") != 0;
 
 #ifndef __TIZEN__
   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);
+      " with pt map %" GST_PTR_FORMAT, stream, pt_map);
 #else
-  GST_LOG_OBJECT (webrtc, "requesting aux sender for stream %" GST_PTR_FORMAT
-      " with transport %" GST_PTR_FORMAT, stream, trans);
+  GST_LOG_OBJECT (webrtc, "requesting aux sender for stream %" GST_PTR_FORMAT, stream);
 #endif
 
   if (have_rtx) {
     GstElement *rtx;
     GstPad *pad;
     gchar *name;
+    GstStructure *merged_local_rtx_ssrc_map =
+        gst_structure_new_empty ("application/x-rtp-ssrc-map");
+    guint i;
 
     if (stream->rtxsend) {
       GST_WARNING_OBJECT (webrtc, "rtprtxsend already created! rtpbin bug?!");
@@ -5336,9 +5344,18 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
     g_object_set (rtx, "max-size-packets", 500, NULL);
     _set_rtx_ptmap_from_stream (webrtc, stream);
 
-    if (WEBRTC_TRANSCEIVER (trans)->local_rtx_ssrc_map)
-      g_object_set (rtx, "ssrc-map",
-          WEBRTC_TRANSCEIVER (trans)->local_rtx_ssrc_map, NULL);
+    for (i = 0; i < webrtc->priv->transceivers->len; i++) {
+      WebRTCTransceiver *trans =
+          WEBRTC_TRANSCEIVER (g_ptr_array_index (webrtc->priv->transceivers,
+              i));
+
+      if (trans->stream == stream && trans->local_rtx_ssrc_map)
+        gst_structure_foreach (trans->local_rtx_ssrc_map,
+            _merge_structure, merged_local_rtx_ssrc_map);
+    }
+
+    g_object_set (rtx, "ssrc-map", merged_local_rtx_ssrc_map, NULL);
+    gst_structure_free (merged_local_rtx_ssrc_map);
 
     gst_bin_add (GST_BIN (ret), rtx);
 
-- 
2.34.1