Update RTPSender::IsFecPacket for FlexFEC.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 769 matching lines @@
     CountPacket(&counters->retransmitted, packet);
     nack_bitrate_sent_.Update(packet.size(), now_ms);
   }
   CountPacket(&counters->transmitted, packet);
 
   if (rtp_stats_callback_)
     rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
 }
 
 bool RTPSender::IsFecPacket(const RtpPacketToSend& packet) const {
-  if (!video_) {
+  if (!video_)
     return false;
-  }
+
+  // FlexFEC.
+  if (packet.Ssrc() == FlexfecSsrc())
+    return true;
+
+  // RED+ULPFEC.
   int pt_red;
   int pt_fec;
   video_->GetUlpfecConfig(&pt_red, &pt_fec);
-  const bool fec_enabled = (pt_fec != -1);
-  return fec_enabled && static_cast<int>(packet.PayloadType()) == pt_red &&
+  return static_cast<int>(packet.PayloadType()) == pt_red &&
          static_cast<int>(packet.payload()[0]) == pt_fec;
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
   if (audio_configured_ || bytes == 0)
     return 0;
   size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
   if (bytes_sent < bytes)
     bytes_sent += SendPadData(bytes - bytes_sent, probe_cluster_id);
   return bytes_sent;
@@ skipping 430 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc