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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/video/video_timing.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/video/video_timing.h | 132 |
1 files changed, 132 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/video/video_timing.h b/third_party/libwebrtc/api/video/video_timing.h new file mode 100644 index 0000000000..698477a81a --- /dev/null +++ b/third_party/libwebrtc/api/video/video_timing.h @@ -0,0 +1,132 @@ +/* + * 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" + +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 { + 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 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. +// +// A value < 0 indicates no change from previous valid value. +// +// 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. +struct VideoPlayoutDelay { + VideoPlayoutDelay() = default; + VideoPlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {} + int min_ms = -1; + int max_ms = -1; + + bool operator==(const VideoPlayoutDelay& rhs) const { + return min_ms == rhs.min_ms && max_ms == rhs.max_ms; + } +}; + +// TODO(bugs.webrtc.org/7660): Old name, delete after downstream use is updated. +using PlayoutDelay = VideoPlayoutDelay; + +} // namespace webrtc + +#endif // API_VIDEO_VIDEO_TIMING_H_ |