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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 <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_