Make FlexFEC packets paceable through RTPSender.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 211 matching lines @@
 
 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
   rtp_sender_.SetRtxSsrc(ssrc);
 }
 
 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
                                               int associated_payload_type) {
   rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type);
 }
 
+rtc::Optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
+  return rtp_sender_.FlexfecSsrc();
+}
+
 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
     const uint8_t* rtcp_packet,
     const size_t length) {
   return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
     const CodecInst& voice_codec) {
   return rtp_sender_.RegisterPayload(
       voice_codec.plname, voice_codec.pltype, voice_codec.plfreq,
@@ skipping 151 matching lines @@
   return rtp_sender_.SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
       payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
                                          uint16_t sequence_number,
                                          int64_t capture_time_ms,
                                          bool retransmission,
                                          int probe_cluster_id) {
-  if (SendingMedia() && ssrc == rtp_sender_.SSRC()) {
-    return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms,
+  if (!SendingMedia()) {
+    return true;
+  }
+  uint32_t media_ssrc = rtp_sender_.SSRC();
+  rtc::Optional<uint32_t> flexfec_ssrc = rtp_sender_.FlexfecSsrc();
+  if (ssrc == media_ssrc || (flexfec_ssrc && ssrc == flexfec_ssrc)) {
+    return rtp_sender_.TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
                                         retransmission, probe_cluster_id);
   }
   // No RTP sender is interested in sending this packet.
   return true;
 }
 
 size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes,
                                             int probe_cluster_id) {
   return rtp_sender_.TimeToSendPadding(bytes, probe_cluster_id);
 }
@@ skipping 530 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc