Implement timing frames.

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_header_extensions.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 RtpPacketToSend& packet) = default;
   RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
       : Packet(extensions, capacity) {}
 
   RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default;
+
   // 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; }
 
+  void set_packetization_finish_time_ms(int64_t time) {
+    SetExtension<VideoTimingExtension>(
+        VideoTiming::GetDeltaMs(capture_time_ms_, time),
+        VideoTiming::kPacketizationFinishDeltaIdx);
+  }
+
+  void set_pacer_exit_time_ms(int64_t time) {
+    SetExtension<VideoTimingExtension>(
+        VideoTiming::GetDeltaMs(capture_time_ms_, time),
+        VideoTiming::kPacerExitDeltaIdx);
+  }
+
+  void set_network_time_ms(int64_t time) {
+    SetExtension<VideoTimingExtension>(
+        VideoTiming::GetDeltaMs(capture_time_ms_, time),
+        VideoTiming::kNetworkTimestampDeltaIdx);
+  }
+
  private:
   int64_t capture_time_ms_ = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_