1 /* 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef API_VIDEO_VIDEO_TIMING_H_ 12 #define API_VIDEO_VIDEO_TIMING_H_ 13 14 #include <stdint.h> 15 16 #include <limits> 17 #include <string> 18 19 namespace webrtc { 20 21 // Video timing timestamps in ms counted from capture_time_ms of a frame. 22 // This structure represents data sent in video-timing RTP header extension. 23 struct VideoSendTiming { 24 enum TimingFrameFlags : uint8_t { 25 kNotTriggered = 0, // Timing info valid, but not to be transmitted. 26 // Used on send-side only. 27 kTriggeredByTimer = 1 << 0, // Frame marked for tracing by periodic timer. 28 kTriggeredBySize = 1 << 1, // Frame marked for tracing due to size. 29 kInvalid = std::numeric_limits<uint8_t>::max() // Invalid, ignore! 30 }; 31 32 // Returns |time_ms - base_ms| capped at max 16-bit value. 33 // Used to fill this data structure as per 34 // https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores 35 // 16-bit deltas of timestamps from packet capture time. 36 static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms); 37 38 uint16_t encode_start_delta_ms; 39 uint16_t encode_finish_delta_ms; 40 uint16_t packetization_finish_delta_ms; 41 uint16_t pacer_exit_delta_ms; 42 uint16_t network_timestamp_delta_ms; 43 uint16_t network2_timestamp_delta_ms; 44 uint8_t flags; 45 }; 46 47 // Used to report precise timings of a 'timing frames'. Contains all important 48 // timestamps for a lifetime of that specific frame. Reported as a string via 49 // GetStats(). Only frame which took the longest between two GetStats calls is 50 // reported. 51 struct TimingFrameInfo { 52 TimingFrameInfo(); 53 54 // Returns end-to-end delay of a frame, if sender and receiver timestamps are 55 // synchronized, -1 otherwise. 56 int64_t EndToEndDelay() const; 57 58 // Returns true if current frame took longer to process than |other| frame. 59 // If other frame's clocks are not synchronized, current frame is always 60 // preferred. 61 bool IsLongerThan(const TimingFrameInfo& other) const; 62 63 // Returns true if flags are set to indicate this frame was marked for tracing 64 // due to the size being outside some limit. 65 bool IsOutlier() const; 66 67 // Returns true if flags are set to indicate this frame was marked fro tracing 68 // due to cyclic timer. 69 bool IsTimerTriggered() const; 70 71 // Returns true if the timing data is marked as invalid, in which case it 72 // should be ignored. 73 bool IsInvalid() const; 74 75 std::string ToString() const; 76 77 bool operator<(const TimingFrameInfo& other) const; 78 79 bool operator<=(const TimingFrameInfo& other) const; 80 81 uint32_t rtp_timestamp; // Identifier of a frame. 82 // All timestamps below are in local monotonous clock of a receiver. 83 // If sender clock is not yet estimated, sender timestamps 84 // (capture_time_ms ... pacer_exit_ms) are negative values, still 85 // relatively correct. 86 int64_t capture_time_ms; // Captrue time of a frame. 87 int64_t encode_start_ms; // Encode start time. 88 int64_t encode_finish_ms; // Encode completion time. 89 int64_t packetization_finish_ms; // Time when frame was passed to pacer. 90 int64_t pacer_exit_ms; // Time when last packet was pushed out of pacer. 91 // Two in-network RTP processor timestamps: meaning is application specific. 92 int64_t network_timestamp_ms; 93 int64_t network2_timestamp_ms; 94 int64_t receive_start_ms; // First received packet time. 95 int64_t receive_finish_ms; // Last received packet time. 96 int64_t decode_start_ms; // Decode start time. 97 int64_t decode_finish_ms; // Decode completion time. 98 int64_t render_time_ms; // Proposed render time to insure smooth playback. 99 100 uint8_t flags; // Flags indicating validity and/or why tracing was triggered. 101 }; 102 103 // Minimum and maximum playout delay values from capture to render. 104 // These are best effort values. 105 // 106 // A value < 0 indicates no change from previous valid value. 107 // 108 // min = max = 0 indicates that the receiver should try and render 109 // frame as soon as possible. 110 // 111 // min = x, max = y indicates that the receiver is free to adapt 112 // in the range (x, y) based on network jitter. 113 struct VideoPlayoutDelay { 114 VideoPlayoutDelay() = default; VideoPlayoutDelayVideoPlayoutDelay115 VideoPlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {} 116 int min_ms = -1; 117 int max_ms = -1; 118 119 bool operator==(const VideoPlayoutDelay& rhs) const { 120 return min_ms == rhs.min_ms && max_ms == rhs.max_ms; 121 } 122 }; 123 124 // TODO(bugs.webrtc.org/7660): Old name, delete after downstream use is updated. 125 using PlayoutDelay = VideoPlayoutDelay; 126 127 } // namespace webrtc 128 129 #endif // API_VIDEO_VIDEO_TIMING_H_ 130