Implement timing frames.

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 "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 {
+  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;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_VIDEO_VIDEO_TIMING_H_