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 <string>

#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/safe_conversions.h"

namespace webrtc {

// 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_