Report timing frames info in GetStats.

webrtc/api/video/video_timing.h

 /*
  *  Copyright (c) 2017 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_API_VIDEO_VIDEO_TIMING_H_
 #define WEBRTC_API_VIDEO_VIDEO_TIMING_H_
 
 #include <stdint.h>
-#include <limits>
+
+#include <string>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 
 namespace webrtc {
 
-// Video timing timstamps in ms counted from capture_time_ms of a frame.
-struct VideoTiming {
+// Video timing timestamps in ms counted from capture_time_ms of a frame.
+// This structure represents data sent in video-timing RTP header extension.
+struct VideoSendTiming {
   static const uint8_t kEncodeStartDeltaIdx = 0;
   static const uint8_t kEncodeFinishDeltaIdx = 1;
   static const uint8_t kPacketizationFinishDeltaIdx = 2;
   static const uint8_t kPacerExitDeltaIdx = 3;
   static const uint8_t kNetworkTimestampDeltaIdx = 4;
   static const uint8_t kNetwork2TimestampDeltaIdx = 5;
 
   // Returns |time_ms - base_ms| capped at max 16-bit value.
   // Used to fill this data structure as per
   // https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores
   // 16-bit deltas of timestamps from packet capture time.
   static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) {
     RTC_DCHECK_GE(time_ms, base_ms);
     return rtc::saturated_cast<uint16_t>(time_ms - base_ms);
   }
 
   uint16_t encode_start_delta_ms;
   uint16_t encode_finish_delta_ms;
   uint16_t packetization_finish_delta_ms;
   uint16_t pacer_exit_delta_ms;
   uint16_t network_timstamp_delta_ms;
   uint16_t network2_timstamp_delta_ms;
   bool is_timing_frame;
 };
 
+// Used to report precise timings of a 'timing frames'. Contains all important
+// timestamps for a lifetime of that specific frame. Reported as a string via
+// GetStats(). Only frame which took the longest between two GetStats calls is
+// reported.
+struct TimingFrameInfo {
+  TimingFrameInfo();
+
+  // Returns end-to-end delay of a frame, if sender and receiver timestamps are
+  // synchronized, -1 otherwise.
+  int64_t EndToEndDelay() const;
+
+  // Returns true if current frame took longer to process than |other| frame.
+  // If other frame's clocks are not synchronized, current frame is always
+  // preferred.
+  bool IsLongerThan(const TimingFrameInfo& other) const;
+
+  std::string ToString() const;
+
+  uint32_t rtp_timestamp;  // Identifier of a frame.
+  // All timestamps below are in local monotonous clock of a receiver.
+  // If sender clock is not yet estimated, sender timestamps
+  // (capture_time_ms ... pacer_exit_ms) are negative values, still
+  // relatively correct.
+  int64_t capture_time_ms;          // Captrue time of a frame.
+  int64_t encode_start_ms;          // Encode start time.
+  int64_t encode_finish_ms;         // Encode completion time.
+  int64_t packetization_finish_ms;  // Time when frame was passed to pacer.
+  int64_t pacer_exit_ms;  // Time when last packet was pushed out of pacer.
+  // Two in-network RTP processor timestamps: meaning is application specific.
+  int64_t network_timestamp_ms;
+  int64_t network2_timestamp_ms;
+  int64_t receive_start_ms;   // First received packet time.
+  int64_t receive_finish_ms;  // Last received packet time.
+  int64_t decode_start_ms;    // Decode start time.
+  int64_t decode_finish_ms;   // Decode completion time.
+  int64_t render_time_ms;     // Proposed render time to insure smooth playback.
+};
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_VIDEO_VIDEO_TIMING_H_