Adjust RtcpReceiver to be testable with callbacks:

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 <list>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gtest_prod_util.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 {
+class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::SenderCallbacks {
  public:
   explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
 
   // Returns the number of milliseconds until the module want a worker thread to
   // call Process.
   int64_t TimeUntilNextProcess() override;
 
   // Process any pending tasks such as timeouts.
   void Process() override;
 
@@ skipping 157 matching lines @@
   void SetREMBStatus(bool enable) override;
 
   void SetREMBData(uint32_t bitrate,
                    const std::vector<uint32_t>& ssrcs) override;
 
   // (TMMBR) Temporary Max Media Bit Rate.
   bool TMMBR() const override;
 
   void SetTMMBRStatus(bool enable) override;
 
-  void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set);
+  void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
 
   uint16_t MaxPayloadLength() const override;
 
   uint16_t MaxDataPayloadLength() const override;
 
   int32_t SetMaxTransferUnit(uint16_t size) override;
 
   int32_t SetTransportOverhead(bool tcp,
                                bool ipv6,
                                uint8_t authentication_overhead = 0) override;
@@ skipping 86 matching lines @@
                    uint32_t* nackRate) const override;
 
   // Good state of RTP receiver inform sender.
   int32_t SendRTCPReferencePictureSelection(uint64_t picture_id) override;
 
   void RegisterSendChannelRtpStatisticsCallback(
       StreamDataCountersCallback* callback) override;
   StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
       const override;
 
-  void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
-  void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
-
-  void OnRequestSendReport();
+  void OnReceivedNACK(
+      const std::list<uint16_t>& nack_sequence_numbers) override;
+  void OnReceivedRtcpReportBlocks(
+      const ReportBlockList& report_blocks) override;
+  void OnRequestSendReport() override;
 
  protected:
   bool UpdateRTCPReceiveInformationTimers();
 
   RTPSender rtp_sender_;
 
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   Clock* clock_;
@@ skipping 31 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_