Add support for transport wide sequence numbers

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 <map>
 
 #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/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-class RTPFragmentationHeader;
 class RtpRtcp;
-struct RTPVideoHeader;
 
 // 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);
+  void OnSsrcChanged();
 
   // 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();
+
  private:
-  // TODO(holmer): When the new video API has launched, remove crit_ and
-  // assume rtp_modules_ will never change during a call. We should then also
-  // switch rtp_modules_ to a map from ssrc to rtp module.
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  void UpdateModuleMap() EXCLUSIVE_LOCKS_REQUIRED(modules_lock_);
 
+  rtc::CriticalSection modules_lock_;
   // Map from ssrc to sending rtp module.
-  std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
+  std::map<uint32_t, RtpRtcp*> rtp_modules_ GUARDED_BY(modules_lock_);
+  volatile int dirty_map_;
+
+  volatile int transport_seq_;
 
   DISALLOW_COPY_AND_ASSIGN(PacketRouter);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_