Propagate packet pacing information to SenTimeHistory

webrtc/modules/rtp_rtcp/include/rtp_rtcp.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 231 matching lines @@
                                 const uint8_t* payload_data,
                                 size_t payload_size,
                                 const RTPFragmentationHeader* fragmentation,
                                 const RTPVideoHeader* rtp_video_header,
                                 uint32_t* transport_frame_id_out) = 0;
 
   virtual bool TimeToSendPacket(uint32_t ssrc,
                                 uint16_t sequence_number,
                                 int64_t capture_time_ms,
                                 bool retransmission,
-                                int probe_cluster_id) = 0;
+                                const PacedPacketInfo& pacing_info) = 0;
 
-  virtual size_t TimeToSendPadding(size_t bytes, int probe_cluster_id) = 0;
+  virtual size_t TimeToSendPadding(size_t bytes,
+                                   const PacedPacketInfo& pacing_info) = 0;
 
   // Called on generation of new statistics after an RTP send.
   virtual void RegisterSendChannelRtpStatisticsCallback(
       StreamDataCountersCallback* callback) = 0;
   virtual StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
       const = 0;
 
   // **************************************************************************
   // RTCP
   // **************************************************************************
@@ skipping 207 matching lines @@
   virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
 
   // Sends a request for a keyframe.
   // Returns -1 on failure else 0.
   virtual int32_t RequestKeyFrame() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_