Make FlexFEC packets paceable through RTPSender.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gtest_prod_util.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 namespace webrtc {
 
 class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
@@ skipping 60 matching lines @@
   RTCPSender::FeedbackState GetFeedbackState();
 
   void SetRtxSendStatus(int mode) override;
   int RtxSendStatus() const override;
 
   void SetRtxSsrc(uint32_t ssrc) override;
 
   void SetRtxSendPayloadType(int payload_type,
                              int associated_payload_type) override;
 
+  rtc::Optional<uint32_t> FlexfecSsrc() const override;
+
   // Sends kRtcpByeCode when going from true to false.
   int32_t SetSendingStatus(bool sending) override;
 
   bool Sending() const override;
 
   // Drops or relays media packets.
   void SetSendingMediaStatus(bool sending) override;
 
   bool SendingMedia() const override;
 
@@ skipping 243 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_