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Searched refs:parsed_log (Results 1 – 13 of 13) sorted by relevance

/external/webrtc/rtc_tools/rtc_event_log_visualizer/
Danalyzer_common.cc16 bool IsRtxSsrc(const ParsedRtcEventLog& parsed_log, in IsRtxSsrc() argument
20 return parsed_log.incoming_rtx_ssrcs().find(ssrc) != in IsRtxSsrc()
21 parsed_log.incoming_rtx_ssrcs().end(); in IsRtxSsrc()
23 return parsed_log.outgoing_rtx_ssrcs().find(ssrc) != in IsRtxSsrc()
24 parsed_log.outgoing_rtx_ssrcs().end(); in IsRtxSsrc()
28 bool IsVideoSsrc(const ParsedRtcEventLog& parsed_log, in IsVideoSsrc() argument
32 return parsed_log.incoming_video_ssrcs().find(ssrc) != in IsVideoSsrc()
33 parsed_log.incoming_video_ssrcs().end(); in IsVideoSsrc()
35 return parsed_log.outgoing_video_ssrcs().find(ssrc) != in IsVideoSsrc()
36 parsed_log.outgoing_video_ssrcs().end(); in IsVideoSsrc()
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Danalyze_audio.h27 void CreateAudioEncoderTargetBitrateGraph(const ParsedRtcEventLog& parsed_log,
30 void CreateAudioEncoderFrameLengthGraph(const ParsedRtcEventLog& parsed_log,
33 void CreateAudioEncoderPacketLossGraph(const ParsedRtcEventLog& parsed_log,
36 void CreateAudioEncoderEnableFecGraph(const ParsedRtcEventLog& parsed_log,
39 void CreateAudioEncoderEnableDtxGraph(const ParsedRtcEventLog& parsed_log,
42 void CreateAudioEncoderNumChannelsGraph(const ParsedRtcEventLog& parsed_log,
48 NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
53 void CreateAudioJitterBufferGraph(const ParsedRtcEventLog& parsed_log,
59 const ParsedRtcEventLog& parsed_log,
66 const ParsedRtcEventLog& parsed_log,
Dalerts.cc38 void TriageHelper::AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log, in AnalyzeStreamGaps() argument
65 const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us(); in AnalyzeStreamGaps()
68 for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) { in AnalyzeStreamGaps()
69 if (IsRtxSsrc(parsed_log, direction, stream.ssrc)) { in AnalyzeStreamGaps()
109 void TriageHelper::AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log, in AnalyzeTransmissionGaps() argument
133 const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us(); in AnalyzeTransmissionGaps()
138 for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) { in AnalyzeTransmissionGaps()
159 for (const auto& rtcp : parsed_log.incoming_rtcp_packets()) { in AnalyzeTransmissionGaps()
173 for (const auto& rtcp : parsed_log.outgoing_rtcp_packets()) { in AnalyzeTransmissionGaps()
193 void TriageHelper::AnalyzeLog(const ParsedRtcEventLog& parsed_log) { in AnalyzeLog() argument
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Dmain.cc251 webrtc::ParsedRtcEventLog parsed_log(header_extensions, in main() local
256 auto status = parsed_log.ParseFile(filename); in main()
268 config.begin_time_ = parsed_log.first_timestamp(); in main()
269 config.end_time_ = parsed_log.last_timestamp(); in main()
277 webrtc::EventLogAnalyzer analyzer(parsed_log, config); in main()
432 CreateAudioEncoderTargetBitrateGraph(parsed_log, config, plot); in main()
435 CreateAudioEncoderFrameLengthGraph(parsed_log, config, plot); in main()
438 CreateAudioEncoderPacketLossGraph(parsed_log, config, plot); in main()
441 CreateAudioEncoderEnableFecGraph(parsed_log, config, plot); in main()
444 CreateAudioEncoderEnableDtxGraph(parsed_log, config, plot); in main()
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Danalyze_audio.cc28 void CreateAudioEncoderTargetBitrateGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderTargetBitrateGraph() argument
45 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderTargetBitrateGraph()
53 void CreateAudioEncoderFrameLengthGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderFrameLengthGraph() argument
70 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderFrameLengthGraph()
78 void CreateAudioEncoderPacketLossGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderPacketLossGraph() argument
95 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderPacketLossGraph()
104 void CreateAudioEncoderEnableFecGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderEnableFecGraph() argument
121 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderEnableFecGraph()
129 void CreateAudioEncoderEnableDtxGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderEnableDtxGraph() argument
146 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderEnableDtxGraph()
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Dalerts.h52 void AnalyzeLog(const ParsedRtcEventLog& parsed_log);
54 void AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
56 void AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
Danalyzer_common.h71 bool IsRtxSsrc(const ParsedRtcEventLog& parsed_log,
74 bool IsVideoSsrc(const ParsedRtcEventLog& parsed_log,
77 bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
81 std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
/external/webrtc/modules/audio_coding/neteq/tools/
Drtc_event_log_source.cc44 ParsedRtcEventLog parsed_log; in CreateFromFile() local
45 auto status = parsed_log.ParseFile(file_name); in CreateFromFile()
51 if (!source->Initialize(parsed_log, ssrc_filter)) { in CreateFromFile()
63 ParsedRtcEventLog parsed_log; in CreateFromString() local
64 auto status = parsed_log.ParseString(file_contents); in CreateFromString()
70 if (!source->Initialize(parsed_log, ssrc_filter)) { in CreateFromString()
98 bool RtcEventLogSource::Initialize(const ParsedRtcEventLog& parsed_log, in Initialize() argument
101 parsed_log.stop_log_events().empty() in Initialize()
103 : parsed_log.stop_log_events().front().log_time_us(); in Initialize()
134 for (const auto& rtp_packets : parsed_log.incoming_rtp_packets_by_ssrc()) { in Initialize()
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Drtc_event_log_source.h56 bool Initialize(const ParsedRtcEventLog& parsed_log,
/external/webrtc/logging/rtc_event_log/
Drtc_event_log_unittest.cc547 ParsedRtcEventLog parsed_log; in ReadAndVerifyLog() local
548 ASSERT_TRUE(parsed_log.ParseFile(temp_filename_).ok()); in ReadAndVerifyLog()
551 auto& parsed_start_log_events = parsed_log.start_log_events(); in ReadAndVerifyLog()
556 auto& parsed_stop_log_events = parsed_log.stop_log_events(); in ReadAndVerifyLog()
560 auto& parsed_alr_state_events = parsed_log.alr_state_events(); in ReadAndVerifyLog()
566 auto& parsed_route_change_events = parsed_log.route_change_events(); in ReadAndVerifyLog()
573 const auto& parsed_audio_playout_map = parsed_log.audio_playout_events(); in ReadAndVerifyLog()
587 parsed_log.audio_network_adaptation_events(); in ReadAndVerifyLog()
595 auto& parsed_bwe_delay_updates = parsed_log.bwe_delay_updates(); in ReadAndVerifyLog()
602 auto& parsed_bwe_loss_updates = parsed_log.bwe_loss_updates(); in ReadAndVerifyLog()
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Drtc_event_log_parser.h900 const ParsedRtcEventLog& parsed_log); in RTC_POP_IGNORING_WUNDEF()
Drtc_event_log_parser.cc2299 const ParsedRtcEventLog& parsed_log) { in GetNetworkTrace() argument
2302 parsed_log.GetPacketInfos(PacketDirection::kOutgoingPacket)) { in GetNetworkTrace()
/external/webrtc/logging/rtc_event_log/encoder/
Drtc_event_log_encoder_unittest.cc77 const ParsedRtcEventLog* parsed_log,
131 RtcEventLogEncoderTest::GetRtpPacketsBySsrc(const ParsedRtcEventLog* parsed_log, in GetRtpPacketsBySsrc() argument
133 const auto& incoming_streams = parsed_log->incoming_rtp_packets_by_ssrc(); in GetRtpPacketsBySsrc()
144 RtcEventLogEncoderTest::GetRtpPacketsBySsrc(const ParsedRtcEventLog* parsed_log, in GetRtpPacketsBySsrc() argument
146 const auto& outgoing_streams = parsed_log->outgoing_rtp_packets_by_ssrc(); in GetRtpPacketsBySsrc()