Let PacketRouter separate send and receive modules.

webrtc/modules/pacing/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.
  */
 
@@ skipping 11 matching lines @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class RtpRtcp;
 namespace rtcp {
 class TransportFeedback;
 }  // namespace rtcp
 
 // PacketRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.

[the sun, 2017/03/31 07:17:13]

This comment doesn't make sense to me. The class "seems" to be used for audio as
well. There is no reference to RTPVideoHeader in this file or in the audio
stream classes.

[nisse-webrtc, 2017/03/31 09:41:32]

I'm rewriting the comment. Stefan, do you know the history, and when simulcast
logic was moved elsewhere?

 class PacketRouter : public PacedSender::PacketSender,
                      public TransportSequenceNumberAllocator {
  public:
   PacketRouter();
   virtual ~PacketRouter();
 
-  void AddRtpModule(RtpRtcp* rtp_module);
-  void RemoveRtpModule(RtpRtcp* rtp_module);
+  // TODO(nisse): Delete, as soon as downstream app is updated.
+  RTC_DEPRECATED void AddRtpModule(RtpRtcp* rtp_module) {
+    AddReceiveRtpModule(rtp_module);
+  }
+  RTC_DEPRECATED void RemoveRtpModule(RtpRtcp* rtp_module) {
+    RemoveReceiveRtpModule(rtp_module);
+  }
+  void AddSendRtpModule(RtpRtcp* rtp_module);
+  void RemoveSendRtpModule(RtpRtcp* rtp_module);
+
+  void AddReceiveRtpModule(RtpRtcp* rtp_module);
+  void RemoveReceiveRtpModule(RtpRtcp* rtp_module);
 
   // Implements PacedSender::Callback.
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_timestamp,
                         bool retransmission,
                         const PacedPacketInfo& packet_info) override;
 
   size_t TimeToSendPadding(size_t bytes,
                            const PacedPacketInfo& packet_info) override;
 
   void SetTransportWideSequenceNumber(uint16_t sequence_number);
   uint16_t AllocateSequenceNumber() override;
 
   // Send transport feedback packet to send-side.
   virtual bool SendFeedback(rtcp::TransportFeedback* packet);
 
  private:
   rtc::ThreadChecker pacer_thread_checker_;
   rtc::CriticalSection modules_crit_;
-  std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(modules_crit_);
+  std::list<RtpRtcp*> rtp_send_modules_ GUARDED_BY(modules_crit_);
+  std::list<RtpRtcp*> rtp_receive_modules_ GUARDED_BY(modules_crit_);
 
   volatile int transport_seq_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_PACING_PACKET_ROUTER_H_