[Draft] RtpPacket sketched.

webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

 /*
  *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet.h"
 
 namespace webrtc {
 // Class to hold rtp packet with metadata for sender side.
 class RtpPacketToSend : public rtp::Packet {
  public:
   explicit RtpPacketToSend(const ExtensionManager* extensions)
       : Packet(extensions) {}
   RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
       : Packet(extensions, capacity) {}
 
   // Time in local time base as close as it can to frame capture time.
   int64_t capture_time_ms() const { return capture_time_ms_; }
   void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
 
+  int64_t send_time_ms() const { return send_time_ms_; }
+  void set_send_time_ms(int64_t time) { send_time_ms_ = time; }
+
  private:
   int64_t capture_time_ms_ = 0;
+  int64_t send_time_ms_ = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_