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| 1 /* | 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
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| 93 | 93 |
| 94 void CreateNetworkDelayFeedbackGraph(Plot* plot); | 94 void CreateNetworkDelayFeedbackGraph(Plot* plot); |
| 95 void CreateTimestampGraph(Plot* plot); | 95 void CreateTimestampGraph(Plot* plot); |
| 96 | 96 |
| 97 void CreateAudioEncoderTargetBitrateGraph(Plot* plot); | 97 void CreateAudioEncoderTargetBitrateGraph(Plot* plot); |
| 98 void CreateAudioEncoderFrameLengthGraph(Plot* plot); | 98 void CreateAudioEncoderFrameLengthGraph(Plot* plot); |
| 99 void CreateAudioEncoderUplinkPacketLossFractionGraph(Plot* plot); | 99 void CreateAudioEncoderUplinkPacketLossFractionGraph(Plot* plot); |
| 100 void CreateAudioEncoderEnableFecGraph(Plot* plot); | 100 void CreateAudioEncoderEnableFecGraph(Plot* plot); |
| 101 void CreateAudioEncoderEnableDtxGraph(Plot* plot); | 101 void CreateAudioEncoderEnableDtxGraph(Plot* plot); |
| 102 void CreateAudioEncoderNumChannelsGraph(Plot* plot); | 102 void CreateAudioEncoderNumChannelsGraph(Plot* plot); |
| 103 void CreateAudioJitterBufferGraph(const std::string& replacement_file_name, | |
| 104 int file_sample_rate_hz, | |
| 105 Plot* plot); | |
| 103 | 106 |
| 104 // Returns a vector of capture and arrival timestamps for the video frames | 107 // Returns a vector of capture and arrival timestamps for the video frames |
| 105 // of the stream with the most number of frames. | 108 // of the stream with the most number of frames. |
| 106 std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const; | 109 std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const; |
| 107 | 110 |
| 108 private: | 111 private: |
| 109 class StreamId { | 112 class StreamId { |
| 110 public: | 113 public: |
| 111 StreamId(uint32_t ssrc, webrtc::PacketDirection direction) | 114 StreamId(uint32_t ssrc, webrtc::PacketDirection direction) |
| 112 : ssrc_(ssrc), direction_(direction) {} | 115 : ssrc_(ssrc), direction_(direction) {} |
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| 156 std::set<StreamId> video_ssrcs_; | 159 std::set<StreamId> video_ssrcs_; |
| 157 std::set<StreamId> audio_ssrcs_; | 160 std::set<StreamId> audio_ssrcs_; |
| 158 | 161 |
| 159 // Maps a stream identifier consisting of ssrc and direction to the parsed | 162 // Maps a stream identifier consisting of ssrc and direction to the parsed |
| 160 // RTP headers in that stream. Header extensions are parsed if the stream | 163 // RTP headers in that stream. Header extensions are parsed if the stream |
| 161 // has been configured. | 164 // has been configured. |
| 162 std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_; | 165 std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_; |
| 163 | 166 |
| 164 std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_; | 167 std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_; |
| 165 | 168 |
| 169 // Maps an SSRC to the timestamps of parsed audio playout events. | |
| 170 std::map<uint32_t, std::vector<uint64_t>> audio_playout_events_; | |
| 171 | |
| 172 // Stores the timestamps for all log end events found. | |
| 173 std::vector<uint64_t> log_end_events_; | |
|
terelius
2017/06/09 14:48:36
std::vector<std::pair<uint64_t, uint64_t>> log_seg
hlundin-webrtc
2017/06/12 07:15:08
I did something, but I'm not sure what assumptions
terelius
2017/06/13 15:16:26
Yes, we write a LOG_START event whenever we start
hlundin-webrtc
2017/06/14 12:09:12
Done. That seems like a good solution.
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| 174 | |
| 166 // A list of all updates from the send-side loss-based bandwidth estimator. | 175 // A list of all updates from the send-side loss-based bandwidth estimator. |
| 167 std::vector<LossBasedBweUpdate> bwe_loss_updates_; | 176 std::vector<LossBasedBweUpdate> bwe_loss_updates_; |
| 168 | 177 |
| 169 std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_; | 178 std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_; |
| 170 | 179 |
| 171 std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent> | 180 std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent> |
| 172 bwe_probe_cluster_created_events_; | 181 bwe_probe_cluster_created_events_; |
| 173 | 182 |
| 174 std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_; | 183 std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_; |
| 175 | 184 |
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| 187 uint64_t end_time_; | 196 uint64_t end_time_; |
| 188 | 197 |
| 189 // Duration (in seconds) of log file. | 198 // Duration (in seconds) of log file. |
| 190 float call_duration_s_; | 199 float call_duration_s_; |
| 191 }; | 200 }; |
| 192 | 201 |
| 193 } // namespace plotting | 202 } // namespace plotting |
| 194 } // namespace webrtc | 203 } // namespace webrtc |
| 195 | 204 |
| 196 #endif // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ | 205 #endif // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ |
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