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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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105 } | 105 } |
106 | 106 |
107 // Deprecated. Pass an explicit file size limit. | 107 // Deprecated. Pass an explicit file size limit. |
108 bool StartLogging(rtc::PlatformFile platform_file) { | 108 bool StartLogging(rtc::PlatformFile platform_file) { |
109 return StartLogging(platform_file, 10000000); | 109 return StartLogging(platform_file, 10000000); |
110 } | 110 } |
111 | 111 |
112 // Stops logging to file and waits until the thread has finished. | 112 // Stops logging to file and waits until the thread has finished. |
113 virtual void StopLogging() = 0; | 113 virtual void StopLogging() = 0; |
114 | 114 |
115 // Logs configuration information for video receive stream. | 115 // Logs configuration information for a video receive stream. |
116 virtual void LogVideoReceiveStreamConfig( | 116 virtual void LogVideoReceiveStreamConfig( |
117 const rtclog::StreamConfig& config) = 0; | 117 const rtclog::StreamConfig& config) = 0; |
118 | 118 |
119 // Logs configuration information for webrtc::VideoSendStream. | 119 // Logs configuration information for a video send stream. |
120 virtual void LogVideoSendStreamConfig( | 120 virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0; |
121 const webrtc::VideoSendStream::Config& config) = 0; | |
122 | 121 |
123 // Logs configuration information for webrtc::AudioReceiveStream. | 122 // Logs configuration information for webrtc::AudioReceiveStream. |
124 virtual void LogAudioReceiveStreamConfig( | 123 virtual void LogAudioReceiveStreamConfig( |
125 const webrtc::AudioReceiveStream::Config& config) = 0; | 124 const webrtc::AudioReceiveStream::Config& config) = 0; |
126 | 125 |
127 // Logs configuration information for webrtc::AudioSendStream. | 126 // Logs configuration information for webrtc::AudioSendStream. |
128 virtual void LogAudioSendStreamConfig( | 127 virtual void LogAudioSendStreamConfig( |
129 const webrtc::AudioSendStream::Config& config) = 0; | 128 const webrtc::AudioSendStream::Config& config) = 0; |
130 | 129 |
131 // Logs the header of an incoming or outgoing RTP packet. packet_length | 130 // Logs the header of an incoming or outgoing RTP packet. packet_length |
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194 public: | 193 public: |
195 bool StartLogging(const std::string& file_name, | 194 bool StartLogging(const std::string& file_name, |
196 int64_t max_size_bytes) override { | 195 int64_t max_size_bytes) override { |
197 return false; | 196 return false; |
198 } | 197 } |
199 bool StartLogging(rtc::PlatformFile platform_file, | 198 bool StartLogging(rtc::PlatformFile platform_file, |
200 int64_t max_size_bytes) override; | 199 int64_t max_size_bytes) override; |
201 void StopLogging() override {} | 200 void StopLogging() override {} |
202 void LogVideoReceiveStreamConfig( | 201 void LogVideoReceiveStreamConfig( |
203 const rtclog::StreamConfig& config) override {} | 202 const rtclog::StreamConfig& config) override {} |
204 void LogVideoSendStreamConfig( | 203 void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {} |
205 const VideoSendStream::Config& config) override {} | |
206 void LogAudioReceiveStreamConfig( | 204 void LogAudioReceiveStreamConfig( |
207 const AudioReceiveStream::Config& config) override {} | 205 const AudioReceiveStream::Config& config) override {} |
208 void LogAudioSendStreamConfig( | 206 void LogAudioSendStreamConfig( |
209 const AudioSendStream::Config& config) override {} | 207 const AudioSendStream::Config& config) override {} |
210 void LogRtpHeader(PacketDirection direction, | 208 void LogRtpHeader(PacketDirection direction, |
211 MediaType media_type, | 209 MediaType media_type, |
212 const uint8_t* header, | 210 const uint8_t* header, |
213 size_t packet_length) override {} | 211 size_t packet_length) override {} |
214 void LogRtpHeader(PacketDirection direction, | 212 void LogRtpHeader(PacketDirection direction, |
215 MediaType media_type, | 213 MediaType media_type, |
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233 int min_probes, | 231 int min_probes, |
234 int min_bytes) override{}; | 232 int min_bytes) override{}; |
235 void LogProbeResultSuccess(int id, int bitrate_bps) override{}; | 233 void LogProbeResultSuccess(int id, int bitrate_bps) override{}; |
236 void LogProbeResultFailure(int id, | 234 void LogProbeResultFailure(int id, |
237 ProbeFailureReason failure_reason) override{}; | 235 ProbeFailureReason failure_reason) override{}; |
238 }; | 236 }; |
239 | 237 |
240 } // namespace webrtc | 238 } // namespace webrtc |
241 | 239 |
242 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ | 240 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
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