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48b8279813
- Moves AnalyzerConfig and helper functions IsAudioSsrc, IsVideoSsrc, IsRtxSsrc, GetStreamNam and GetLayerName to analyzer_common.h - Moves log_segments() code to rtc_event_log_parser.h - Moves TriageAlert/Notification code to a new file with a couple of minor fixes to make it less spammy. Bug: webrtc:11566 Change-Id: Ib33941d8185f7382fc72ed65768e46015e0320de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174824 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31318}
79 lines
2.6 KiB
C++
79 lines
2.6 KiB
C++
/*
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* Copyright (c) 2020 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 RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_COMMON_H_
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#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_COMMON_H_
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#include <cstdint>
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#include <string>
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#include "logging/rtc_event_log/rtc_event_log_parser.h"
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namespace webrtc {
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class AnalyzerConfig {
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public:
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float GetCallTimeSec(int64_t timestamp_us) const {
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int64_t offset = normalize_time_ ? begin_time_ : 0;
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return static_cast<float>(timestamp_us - offset) / 1000000;
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}
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float CallBeginTimeSec() const { return GetCallTimeSec(begin_time_); }
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float CallEndTimeSec() const { return GetCallTimeSec(end_time_); }
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// Window and step size used for calculating moving averages, e.g. bitrate.
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// The generated data points will be |step_| microseconds apart.
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// Only events occurring at most |window_duration_| microseconds before the
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// current data point will be part of the average.
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int64_t window_duration_;
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int64_t step_;
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// First and last events of the log.
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int64_t begin_time_;
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int64_t end_time_;
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bool normalize_time_;
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};
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struct LayerDescription {
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LayerDescription(uint32_t ssrc, uint8_t spatial_layer, uint8_t temporal_layer)
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: ssrc(ssrc),
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spatial_layer(spatial_layer),
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temporal_layer(temporal_layer) {}
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bool operator<(const LayerDescription& other) const {
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if (ssrc != other.ssrc)
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return ssrc < other.ssrc;
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if (spatial_layer != other.spatial_layer)
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return spatial_layer < other.spatial_layer;
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return temporal_layer < other.temporal_layer;
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}
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uint32_t ssrc;
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uint8_t spatial_layer;
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uint8_t temporal_layer;
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};
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bool IsRtxSsrc(const ParsedRtcEventLog& parsed_log,
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PacketDirection direction,
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uint32_t ssrc);
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bool IsVideoSsrc(const ParsedRtcEventLog& parsed_log,
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PacketDirection direction,
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uint32_t ssrc);
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bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
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PacketDirection direction,
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uint32_t ssrc);
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std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
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PacketDirection direction,
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uint32_t ssrc);
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std::string GetLayerName(LayerDescription layer);
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} // namespace webrtc
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#endif // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_COMMON_H_
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