c20978e581
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL. NOTRY=True NOTREECHECKS=True TBR=kwiberg@webrtc.org, kjellander@webrtc.org Bug: webrtc:7634 Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36 Reviewed-on: https://chromium-review.googlesource.com/562156 Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18919}
92 lines
3.6 KiB
C++
92 lines
3.6 KiB
C++
/*
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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 WEBRTC_API_VIDEO_VIDEO_TIMING_H_
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#define WEBRTC_API_VIDEO_VIDEO_TIMING_H_
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#include <stdint.h>
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#include <string>
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/safe_conversions.h"
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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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static const uint8_t kEncodeStartDeltaIdx = 0;
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static const uint8_t kEncodeFinishDeltaIdx = 1;
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static const uint8_t kPacketizationFinishDeltaIdx = 2;
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static const uint8_t kPacerExitDeltaIdx = 3;
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static const uint8_t kNetworkTimestampDeltaIdx = 4;
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static const uint8_t kNetwork2TimestampDeltaIdx = 5;
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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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RTC_DCHECK_GE(time_ms, base_ms);
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return rtc::saturated_cast<uint16_t>(time_ms - base_ms);
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}
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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_timstamp_delta_ms;
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uint16_t network2_timstamp_delta_ms;
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bool is_timing_frame;
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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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std::string ToString() 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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};
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} // namespace webrtc
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#endif // WEBRTC_API_VIDEO_VIDEO_TIMING_H_
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