| /* |
| * 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 "rtc_base/checks.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| |
| namespace webrtc { |
| |
| enum TimingFrameFlags : uint8_t { |
| kNotTriggered = 0, // Timing info valid, but not to be transmitted. |
| // Used on send-side only. |
| // TODO(ilnik): Delete compatibility alias. |
| // Used to be sent over the wire, for the old protocol. |
| kDefault = 0, // Old name, for API compatibility. |
| 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! |
| }; |
| |
| // 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 { |
| // Offsets of the fields in the RTP header extension, counting from the first |
| // byte after the one-byte header. |
| static constexpr uint8_t kFlagsOffset = 0; |
| static constexpr uint8_t kEncodeStartDeltaOffset = 1; |
| static constexpr uint8_t kEncodeFinishDeltaOffset = 3; |
| static constexpr uint8_t kPacketizationFinishDeltaOffset = 5; |
| static constexpr uint8_t kPacerExitDeltaOffset = 7; |
| static constexpr uint8_t kNetworkTimestampDeltaOffset = 9; |
| static constexpr uint8_t kNetwork2TimestampDeltaOffset = 11; |
| |
| // 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) { |
| RTC_DCHECK_GE(time_ms, base_ms); |
| return rtc::saturated_cast<uint16_t>(time_ms - base_ms); |
| } |
| |
| 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; |
| }; |
| |
| // 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. |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_VIDEO_VIDEO_TIMING_H_ |