Add sender controlled playout delay limits

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 {
  public:
   explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
@@ skipping 279 matching lines @@
 
   // 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 OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks);
 
   void OnRequestSendReport();
 
  protected:
   bool UpdateRTCPReceiveInformationTimers();
 
   RTPSender rtp_sender_;
 
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
@@ skipping 33 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_