Add RemoteEstimatorProxy for capturing receive times

webrtc/modules/pacing/include/packet_router.h

 /*
  *  Copyright (c) 2015 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_PACING_INCLUDE_PACKET_ROUTER_H_
 #define WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
 
 #include <list>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
 
 class RtpRtcp;
 
 // PacketRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
 class PacketRouter : public PacedSender::Callback {
  public:
   PacketRouter();
   virtual ~PacketRouter();
 
   void AddRtpModule(RtpRtcp* rtp_module);
   void RemoveRtpModule(RtpRtcp* rtp_module);
 
   // Implements PacedSender::Callback.
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_timestamp,
                         bool retransmission) override;
 
   size_t TimeToSendPadding(size_t bytes) override;
 
   void SetTransportWideSequenceNumber(uint16_t sequence_number);
   uint16_t AllocateSequenceNumber();
 
+  // Send transport feedback packet to send-side.
+  virtual bool SendFeedback(rtcp::TransportFeedback* packet);

[stefan-webrtc, 2015/09/04 07:31:36]

Isn't PacketRouter actually living on the send side? Feels a bit confusing to
me. A receive shouldn't need a PacketRouter?

[sprang_webrtc, 2015/09/04 13:29:44]

Receive times are collected on a call-level, no in the rtp module, so we don't
have a natural way to send the transport feedback rtcp packet back to the send
side. The packet router will find a suitable rtcp sender in a similar manner to
how i finds a suitable rtp sender for the payload on the send side.

[stefan-webrtc, 2015/09/07 10:12:21]

Ok, maybe that is reasonable. How is it set up with the correct modules?

[sprang_webrtc, 2015/09/07 11:14:35]

Modules that can send rtcp are added to this router even if they don't send rtp
payload. That change probably ended up in one of the other CL's in the
pipeline...

+
  private:
   rtc::CriticalSection modules_lock_;
   // Map from ssrc to sending rtp module.
   std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(modules_lock_);
 
   volatile int transport_seq_;
 
   DISALLOW_COPY_AND_ASSIGN(PacketRouter);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_