Add RemoteEstimatorProxy for capturing receive times

webrtc/modules/rtp_rtcp/interface/rtp_rtcp.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_INTERFACE_RTP_RTCP_H_
 #define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
 
 #include <set>
 #include <vector>
 
 #include "webrtc/modules/interface/module.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 namespace webrtc {
 // Forward declarations.
 class PacedSender;
 class PacketRouter;
 class ReceiveStatistics;
 class RemoteBitrateEstimator;
 class RtpReceiver;
 class Transport;
+namespace rtcp {
+class TransportFeedback;
+}
 
 class RtpRtcp : public Module {
  public:
   struct Configuration {
     Configuration();
 
    /*  id                   - Unique identifier of this RTP/RTCP module object
     *  audio                - True for a audio version of the RTP/RTCP module
     *                         object false will create a video version
     *  clock                - The clock to use to read time. If NULL object
@@ skipping 497 matching lines @@
     virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
 
     // Returns true if the module is configured to store packets.
     virtual bool StorePackets() const = 0;
 
     // Called on receipt of RTCP report block from remote side.
     virtual void RegisterRtcpStatisticsCallback(
         RtcpStatisticsCallback* callback) = 0;
     virtual RtcpStatisticsCallback*
         GetRtcpStatisticsCallback() = 0;
+    // BWE feedback packets.
+    virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
 
     /**************************************************************************
     *
     *   Audio
     *
     ***************************************************************************/
 
     /*
     *   set audio packet size, used to determine when it's time to send a DTMF
     *   packet in silence (CNG)
@@ skipping 75 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_