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106 lines
4.0 KiB
106 lines
4.0 KiB
4 months ago
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/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_VIDEO_VIDEO_TIMING_H_
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#define API_VIDEO_VIDEO_TIMING_H_
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#include <stdint.h>
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#include <limits>
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#include <string>
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namespace webrtc {
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// Video timing timestamps in ms counted from capture_time_ms of a frame.
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// This structure represents data sent in video-timing RTP header extension.
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struct VideoSendTiming {
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enum TimingFrameFlags : uint8_t {
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kNotTriggered = 0, // Timing info valid, but not to be transmitted.
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// Used on send-side only.
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kTriggeredByTimer = 1 << 0, // Frame marked for tracing by periodic timer.
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kTriggeredBySize = 1 << 1, // Frame marked for tracing due to size.
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kInvalid = std::numeric_limits<uint8_t>::max() // Invalid, ignore!
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};
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// Returns |time_ms - base_ms| capped at max 16-bit value.
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// Used to fill this data structure as per
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// https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores
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// 16-bit deltas of timestamps from packet capture time.
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static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms);
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uint16_t encode_start_delta_ms;
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uint16_t encode_finish_delta_ms;
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uint16_t packetization_finish_delta_ms;
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uint16_t pacer_exit_delta_ms;
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uint16_t network_timestamp_delta_ms;
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uint16_t network2_timestamp_delta_ms;
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uint8_t flags;
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};
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// Used to report precise timings of a 'timing frames'. Contains all important
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// timestamps for a lifetime of that specific frame. Reported as a string via
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// GetStats(). Only frame which took the longest between two GetStats calls is
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// reported.
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struct TimingFrameInfo {
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TimingFrameInfo();
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// Returns end-to-end delay of a frame, if sender and receiver timestamps are
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// synchronized, -1 otherwise.
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int64_t EndToEndDelay() const;
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// Returns true if current frame took longer to process than |other| frame.
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// If other frame's clocks are not synchronized, current frame is always
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// preferred.
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bool IsLongerThan(const TimingFrameInfo& other) const;
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// Returns true if flags are set to indicate this frame was marked for tracing
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// due to the size being outside some limit.
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bool IsOutlier() const;
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// Returns true if flags are set to indicate this frame was marked fro tracing
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// due to cyclic timer.
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bool IsTimerTriggered() const;
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// Returns true if the timing data is marked as invalid, in which case it
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// should be ignored.
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bool IsInvalid() const;
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std::string ToString() const;
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bool operator<(const TimingFrameInfo& other) const;
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bool operator<=(const TimingFrameInfo& other) const;
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uint32_t rtp_timestamp; // Identifier of a frame.
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// All timestamps below are in local monotonous clock of a receiver.
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// If sender clock is not yet estimated, sender timestamps
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// (capture_time_ms ... pacer_exit_ms) are negative values, still
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// relatively correct.
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int64_t capture_time_ms; // Captrue time of a frame.
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int64_t encode_start_ms; // Encode start time.
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int64_t encode_finish_ms; // Encode completion time.
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int64_t packetization_finish_ms; // Time when frame was passed to pacer.
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int64_t pacer_exit_ms; // Time when last packet was pushed out of pacer.
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// Two in-network RTP processor timestamps: meaning is application specific.
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int64_t network_timestamp_ms;
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int64_t network2_timestamp_ms;
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int64_t receive_start_ms; // First received packet time.
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int64_t receive_finish_ms; // Last received packet time.
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int64_t decode_start_ms; // Decode start time.
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int64_t decode_finish_ms; // Decode completion time.
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int64_t render_time_ms; // Proposed render time to insure smooth playback.
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uint8_t flags; // Flags indicating validity and/or why tracing was triggered.
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};
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} // namespace webrtc
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#endif // API_VIDEO_VIDEO_TIMING_H_
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