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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 API_VIDEO_VIDEO_TIMING_H_
#define API_VIDEO_VIDEO_TIMING_H_
#include <stdint.h>
#include <limits>
#include <string>
#include "api/units/time_delta.h"
#include "rtc_base/system/rtc_export.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 RTC_EXPORT VideoSendTiming {
enum TimingFrameFlags : uint8_t {
kNotTriggered = 0, // Timing info valid, but not to be transmitted.
// Used on send-side only.
kTriggeredByTimer = 1 << 0, // Frame marked for tracing by periodic timer.
kTriggeredBySize = 1 << 1, // Frame marked for tracing due to size.
kInvalid = std::numeric_limits<uint8_t>::max() // Invalid, ignore!
};
// 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);
static uint16_t GetDeltaCappedMs(TimeDelta delta);
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_timestamp_delta_ms;
uint16_t network2_timestamp_delta_ms;
uint8_t flags = TimingFrameFlags::kInvalid;
};
// 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 RTC_EXPORT 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;
// Returns true if flags are set to indicate this frame was marked for tracing
// due to the size being outside some limit.
bool IsOutlier() const;
// Returns true if flags are set to indicate this frame was marked fro tracing
// due to cyclic timer.
bool IsTimerTriggered() const;
// Returns true if the timing data is marked as invalid, in which case it
// should be ignored.
bool IsInvalid() const;
std::string ToString() const;
bool operator<(const TimingFrameInfo& other) const;
bool operator<=(const TimingFrameInfo& other) 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.
uint8_t flags; // Flags indicating validity and/or why tracing was triggered.
};
// Minimum and maximum playout delay values from capture to render.
// These are best effort values.
//
// min = max = 0 indicates that the receiver should try and render
// frame as soon as possible.
//
// min = x, max = y indicates that the receiver is free to adapt
// in the range (x, y) based on network jitter.
// This class ensures invariant 0 <= min <= max <= kMax.
class RTC_EXPORT VideoPlayoutDelay {
public:
// Maximum supported value for the delay limit.
static constexpr TimeDelta kMax = TimeDelta::Millis(10) * 0xFFF;
// Creates delay limits that indicates receiver should try to render frame
// as soon as possible.
static VideoPlayoutDelay Minimal() {
return VideoPlayoutDelay(TimeDelta::Zero(), TimeDelta::Zero());
}
// Creates valid, but unspecified limits.
VideoPlayoutDelay() = default;
VideoPlayoutDelay(const VideoPlayoutDelay&) = default;
VideoPlayoutDelay& operator=(const VideoPlayoutDelay&) = default;
VideoPlayoutDelay(TimeDelta min, TimeDelta max);
bool Set(TimeDelta min, TimeDelta max);
TimeDelta min() const { return min_; }
TimeDelta max() const { return max_; }
friend bool operator==(const VideoPlayoutDelay& lhs,
const VideoPlayoutDelay& rhs) {
return lhs.min_ == rhs.min_ && lhs.max_ == rhs.max_;
}
private:
TimeDelta min_ = TimeDelta::Zero();
TimeDelta max_ = kMax;
};
} // namespace webrtc
#endif // API_VIDEO_VIDEO_TIMING_H_
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