2edc6845ac
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported. The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process). BUG=webrtc:7594 Review-Url: https://codereview.webrtc.org/2946413002 Cr-Commit-Position: refs/heads/master@{#18909}
53 lines
1.7 KiB
C++
53 lines
1.7 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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#include "webrtc/api/video/video_timing.h"
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#include <sstream>
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namespace webrtc {
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TimingFrameInfo::TimingFrameInfo()
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: rtp_timestamp(0),
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capture_time_ms(-1),
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encode_start_ms(-1),
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encode_finish_ms(-1),
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packetization_finish_ms(-1),
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pacer_exit_ms(-1),
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network_timestamp_ms(-1),
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network2_timestamp_ms(-1),
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receive_start_ms(-1),
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receive_finish_ms(-1),
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decode_start_ms(-1),
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decode_finish_ms(-1),
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render_time_ms(-1) {}
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int64_t TimingFrameInfo::EndToEndDelay() const {
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return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1;
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}
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bool TimingFrameInfo::IsLongerThan(const TimingFrameInfo& other) const {
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int64_t other_delay = other.EndToEndDelay();
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return other_delay == -1 || EndToEndDelay() > other_delay;
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}
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std::string TimingFrameInfo::ToString() const {
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std::stringstream out;
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out << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms
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<< ',' << encode_finish_ms << ',' << packetization_finish_ms << ','
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<< pacer_exit_ms << ',' << network_timestamp_ms << ','
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<< network2_timestamp_ms << ',' << receive_start_ms << ','
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<< receive_finish_ms << ',' << decode_start_ms << ',' << decode_finish_ms
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<< ',' << render_time_ms;
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return out.str();
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}
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} // namespace webrtc
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