tor-browser

video_timing.h (5703B)

---

/*
*  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_