Add support for transport wide sequence numbers

webrtc/modules/pacing/include/packet_router.h

Index: webrtc/modules/pacing/include/packet_router.h
diff --git a/webrtc/modules/pacing/include/packet_router.h b/webrtc/modules/pacing/include/packet_router.h
index c1b332a6bfc6b1ad6e7af0bea989b0803a648c89..bed85cb4f70103e8f87880e549122e8b128ffaf9 100644
--- a/webrtc/modules/pacing/include/packet_router.h
+++ b/webrtc/modules/pacing/include/packet_router.h
@@ -11,20 +11,21 @@
 #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/bitrate_controller/send_time_history.h"

[stefan-webrtc, 2015/07/22 10:49:00]

Maybe this isn't the right place for the send_time_history?

[sprang_webrtc, 2015/07/22 15:11:32]

What would be a better place? modules/remote_bitrate_estimator?

[stefan-webrtc, 2015/07/27 12:13:49]

I'm not sure... Maybe under pacing? We can keep it where it is if you'd like,
but it feels a bit weird since it's only used by the packet router / pacer?

[sprang_webrtc, 2015/07/28 13:29:27]

I've move it around a few times now, trying to fit it together with upcoming
parts of the transport feedback wireup. It ended up in remote_bitrate_estimator.
OK?

 #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.
@@ -35,6 +36,7 @@ class PacketRouter : public PacedSender::Callback {
 
   void AddRtpModule(RtpRtcp* rtp_module);
   void RemoveRtpModule(RtpRtcp* rtp_module);
+  void OnSsrcChanged();
 
   // Implements PacedSender::Callback.
   bool TimeToSendPacket(uint32_t ssrc,
@@ -44,14 +46,34 @@ class PacketRouter : public PacedSender::Callback {
 
   size_t TimeToSendPadding(size_t bytes) override;
 
+  // Enable transport wide sequence number extension headers for outgoing
+  // RTP packets, specified by draft-holmer-rmcat-transport-wide-cc-extensions.
+  // This also enables send time history.
+  void EnableTransportWideFeedback();
+
+  void SetTransportWideSequenceNumber(uint16_t sequence_number);
+
+  // Populate PacketInfo.send_time_ms, by looking up the send time in the
+  // stored history index by sequence number. Returns the number of PacketInfo
+  // instances for which the lookup was successful.
+  size_t PopulateSendTimes(std::vector<PacketInfo>* packet_info);
+
+  uint16_t AllocateSequenceNumber();
+  void OnPacketSent(uint16_t sequence_number, int64_t send_time);
+
  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(ssrc_lookup_lock_);
 
+  rtc::CriticalSection ssrc_lookup_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(ssrc_lookup_lock_);
+  volatile int dirty_map_;
+
+  bool transport_wide_seq_enabled_;
+  volatile int transport_seq_;
+
+  rtc::CriticalSection history_lock_;
+  rtc::scoped_ptr<SendTimeHistory> send_time_history_ GUARDED_BY(history_lock_);
 
   DISALLOW_COPY_AND_ASSIGN(PacketRouter);
 };