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/*
* 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_