Implement timing frames.

webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

Index: webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
index f2ddc8a52e467801cc3ed5e72af5fc1945e96ca2..ad09c6dcb54874c40511dc1cd02ec0ac564ad97d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -10,6 +10,7 @@
 #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 {
@@ -23,10 +24,30 @@ class RtpPacketToSend : public rtp::Packet {
       : 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::GetDeltaCappedMs(capture_time_ms_, time),
+        VideoTiming::kPacketizationFinishDeltaIdx);
+  }
+
+  void set_pacer_exit_time_ms(int64_t time) {
+    SetExtension<VideoTimingExtension>(
+        VideoTiming::GetDeltaCappedMs(capture_time_ms_, time),
+        VideoTiming::kPacerExitDeltaIdx);
+  }
+
+  void set_network_time_ms(int64_t time) {
+    SetExtension<VideoTimingExtension>(
+        VideoTiming::GetDeltaCappedMs(capture_time_ms_, time),
+        VideoTiming::kNetworkTimestampDeltaIdx);
+  }
+
  private:
   int64_t capture_time_ms_ = 0;
 };