Propagate packet pacing information to SendTimeHistory.

webrtc/modules/rtp_rtcp/source/rtp_sender.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.
  */
 
@@ skipping 105 matching lines @@
 
   // RTP header extension
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
   bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) const;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_time_ms,
                         bool retransmission,
-                        int probe_cluster_id);
-  size_t TimeToSendPadding(size_t bytes, int probe_cluster_id);
+                        const PacedPacketInfo& pacing_info);
+  size_t TimeToSendPadding(size_t bytes, const PacedPacketInfo& pacing_info);
 
   // NACK.
   int SelectiveRetransmissions() const;
   int SetSelectiveRetransmissions(uint8_t settings);
   void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
                       int64_t avg_rtt);
 
   void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
 
   bool StorePackets() const;
@@ skipping 69 matching lines @@
 
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
 
  private:
   // Maps capture time in milliseconds to send-side delay in milliseconds.
   // Send-side delay is the difference between transmission time and capture
   // time.
   typedef std::map<int64_t, int> SendDelayMap;
 
-  size_t SendPadData(size_t bytes, int probe_cluster_id);
+  size_t SendPadData(size_t bytes, const PacedPacketInfo& pacing_info);
 
   bool PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
                             bool send_over_rtx,
                             bool is_retransmit,
-                            int probe_cluster_id);
+                            const PacedPacketInfo& pacing_info);
 
   // Return the number of bytes sent.  Note that both of these functions may
   // return a larger value that their argument.
-  size_t TrySendRedundantPayloads(size_t bytes, int probe_cluster_id);
+  size_t TrySendRedundantPayloads(size_t bytes,
+                                  const PacedPacketInfo& pacing_info);
 
   std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
       const RtpPacketToSend& packet);
 
   bool SendPacketToNetwork(const RtpPacketToSend& packet,
                            const PacketOptions& options);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
   void UpdateOnSendPacket(int packet_id,
                           int64_t capture_time_ms,
                           uint32_t ssrc);
 
   bool UpdateTransportSequenceNumber(RtpPacketToSend* packet,
                                      int* packet_id) const;
 
   void UpdateRtpStats(const RtpPacketToSend& packet,
                       bool is_rtx,
                       bool is_retransmit);
   bool IsFecPacket(const RtpPacketToSend& packet) const;
 
   void AddPacketToTransportFeedback(uint16_t packet_id,
                                     const RtpPacketToSend& packet,
-                                    int probe_cluster_id);
+                                    const PacedPacketInfo& pacing_info);
 
   void UpdateRtpOverhead(const RtpPacketToSend& packet);
 
   Clock* const clock_;
   const int64_t clock_delta_ms_;
   Random random_ GUARDED_BY(send_critsect_);
 
   const bool audio_configured_;
   const std::unique_ptr<RTPSenderAudio> audio_;
   const std::unique_ptr<RTPSenderVideo> video_;
@@ skipping 64 matching lines @@
   OverheadObserver* overhead_observer_;
 
   const bool send_side_bwe_with_overhead_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_