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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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98 void CreateAudioEncoderTargetBitrateGraph(Plot* plot); | 98 void CreateAudioEncoderTargetBitrateGraph(Plot* plot); |
99 void CreateAudioEncoderFrameLengthGraph(Plot* plot); | 99 void CreateAudioEncoderFrameLengthGraph(Plot* plot); |
100 void CreateAudioEncoderPacketLossGraph(Plot* plot); | 100 void CreateAudioEncoderPacketLossGraph(Plot* plot); |
101 void CreateAudioEncoderEnableFecGraph(Plot* plot); | 101 void CreateAudioEncoderEnableFecGraph(Plot* plot); |
102 void CreateAudioEncoderEnableDtxGraph(Plot* plot); | 102 void CreateAudioEncoderEnableDtxGraph(Plot* plot); |
103 void CreateAudioEncoderNumChannelsGraph(Plot* plot); | 103 void CreateAudioEncoderNumChannelsGraph(Plot* plot); |
104 void CreateAudioJitterBufferGraph(const std::string& replacement_file_name, | 104 void CreateAudioJitterBufferGraph(const std::string& replacement_file_name, |
105 int file_sample_rate_hz, | 105 int file_sample_rate_hz, |
106 Plot* plot); | 106 Plot* plot); |
107 | 107 |
| 108 void CreateQueueDelayGraph(Plot* plot); |
| 109 |
108 // Returns a vector of capture and arrival timestamps for the video frames | 110 // Returns a vector of capture and arrival timestamps for the video frames |
109 // of the stream with the most number of frames. | 111 // of the stream with the most number of frames. |
110 std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const; | 112 std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const; |
111 | 113 |
112 private: | 114 private: |
113 class StreamId { | 115 class StreamId { |
114 public: | 116 public: |
115 StreamId(uint32_t ssrc, webrtc::PacketDirection direction) | 117 StreamId(uint32_t ssrc, webrtc::PacketDirection direction) |
116 : ssrc_(ssrc), direction_(direction) {} | 118 : ssrc_(ssrc), direction_(direction) {} |
117 bool operator<(const StreamId& other) const { | 119 bool operator<(const StreamId& other) const { |
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179 | 181 |
180 std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_; | 182 std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_; |
181 | 183 |
182 std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent> | 184 std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent> |
183 bwe_probe_cluster_created_events_; | 185 bwe_probe_cluster_created_events_; |
184 | 186 |
185 std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_; | 187 std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_; |
186 | 188 |
187 std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_; | 189 std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_; |
188 | 190 |
| 191 std::vector<ParsedRtcEventLog::BweAckedBitrateEvent> acked_bitrate_events_; |
| 192 |
| 193 std::vector<ParsedRtcEventLog::AlrStateEvent> alr_state_events_; |
| 194 |
| 195 std::vector<ParsedRtcEventLog::PacketQueueTime> packet_queue_time_events_; |
| 196 |
189 // Window and step size used for calculating moving averages, e.g. bitrate. | 197 // Window and step size used for calculating moving averages, e.g. bitrate. |
190 // The generated data points will be |step_| microseconds apart. | 198 // The generated data points will be |step_| microseconds apart. |
191 // Only events occuring at most |window_duration_| microseconds before the | 199 // Only events occuring at most |window_duration_| microseconds before the |
192 // current data point will be part of the average. | 200 // current data point will be part of the average. |
193 uint64_t window_duration_; | 201 uint64_t window_duration_; |
194 uint64_t step_; | 202 uint64_t step_; |
195 | 203 |
196 // First and last events of the log. | 204 // First and last events of the log. |
197 uint64_t begin_time_; | 205 uint64_t begin_time_; |
198 uint64_t end_time_; | 206 uint64_t end_time_; |
199 | 207 |
200 // Duration (in seconds) of log file. | 208 // Duration (in seconds) of log file. |
201 float call_duration_s_; | 209 float call_duration_s_; |
202 }; | 210 }; |
203 | 211 |
204 } // namespace plotting | 212 } // namespace plotting |
205 } // namespace webrtc | 213 } // namespace webrtc |
206 | 214 |
207 #endif // WEBRTC_RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ | 215 #endif // WEBRTC_RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ |
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