mirror of
https://github.com/mollyim/webrtc.git
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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}
83 lines
2.5 KiB
C++
83 lines
2.5 KiB
C++
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/*
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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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#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
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namespace webrtc {
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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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if (direction == kIncomingPacket) {
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return parsed_log.incoming_rtx_ssrcs().find(ssrc) !=
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parsed_log.incoming_rtx_ssrcs().end();
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} else {
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return parsed_log.outgoing_rtx_ssrcs().find(ssrc) !=
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parsed_log.outgoing_rtx_ssrcs().end();
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}
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}
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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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if (direction == kIncomingPacket) {
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return parsed_log.incoming_video_ssrcs().find(ssrc) !=
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parsed_log.incoming_video_ssrcs().end();
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} else {
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return parsed_log.outgoing_video_ssrcs().find(ssrc) !=
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parsed_log.outgoing_video_ssrcs().end();
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}
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}
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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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if (direction == kIncomingPacket) {
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return parsed_log.incoming_audio_ssrcs().find(ssrc) !=
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parsed_log.incoming_audio_ssrcs().end();
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} else {
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return parsed_log.outgoing_audio_ssrcs().find(ssrc) !=
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parsed_log.outgoing_audio_ssrcs().end();
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}
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}
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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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char buffer[200];
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rtc::SimpleStringBuilder name(buffer);
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if (IsAudioSsrc(parsed_log, direction, ssrc)) {
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name << "Audio ";
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} else if (IsVideoSsrc(parsed_log, direction, ssrc)) {
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name << "Video ";
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} else {
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name << "Unknown ";
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}
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if (IsRtxSsrc(parsed_log, direction, ssrc)) {
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name << "RTX ";
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}
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if (direction == kIncomingPacket)
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name << "(In) ";
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else
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name << "(Out) ";
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name << "SSRC " << ssrc;
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return name.str();
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}
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std::string GetLayerName(LayerDescription layer) {
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char buffer[100];
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rtc::SimpleStringBuilder name(buffer);
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name << "SSRC " << layer.ssrc << " sl " << layer.spatial_layer << ", tl "
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<< layer.temporal_layer;
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return name.str();
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}
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} // namespace webrtc
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