Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(297)

Side by Side Diff: webrtc/logging/rtc_event_log/rtc_event_log.h

Issue 2997973002: Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.
Patch Set: Created 3 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 105 matching lines...) Expand 10 before | Expand all | Expand 10 after
116 // Logs configuration information for a video send stream. 116 // Logs configuration information for a video send stream.
117 virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0; 117 virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0;
118 118
119 // Logs configuration information for an audio receive stream. 119 // Logs configuration information for an audio receive stream.
120 virtual void LogAudioReceiveStreamConfig( 120 virtual void LogAudioReceiveStreamConfig(
121 const rtclog::StreamConfig& config) = 0; 121 const rtclog::StreamConfig& config) = 0;
122 122
123 // Logs configuration information for an audio send stream. 123 // Logs configuration information for an audio send stream.
124 virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) = 0; 124 virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) = 0;
125 125
126 // Logs the header of an incoming or outgoing RTP packet. packet_length 126 // Logs the header of an incoming RTP packet. |packet_length|
127 // is the total length of the packet, including both header and payload. 127 // is the total length of the packet, including both header and payload.
128 virtual void LogRtpHeader(PacketDirection direction, 128 virtual void LogIncomingRtpHeader(const uint8_t* header,
danilchap 2017/08/18 11:34:03 can you keep old signature (as deprecated) to avoi
129 const uint8_t* header, 129 size_t packet_length) = 0;
130 size_t packet_length) = 0;
131 130
132 // Same as above but used on the sender side to log packets that are part of 131 // Logs the header of an incoming RTP packet. |packet_length|
133 // a probe cluster. 132 // is the total length of the packet, including both header and payload.
134 virtual void LogRtpHeader(PacketDirection direction, 133 virtual void LogOutgoingRtpHeader(const uint8_t* header,
135 const uint8_t* header, 134 size_t packet_length,
136 size_t packet_length, 135 int probe_cluster_id) = 0;
137 int probe_cluster_id) = 0;
138 136
139 // Logs an incoming or outgoing RTCP packet. 137 // Logs an incoming RTCP packet.
140 virtual void LogRtcpPacket(PacketDirection direction, 138 virtual void LogIncomingRtcpPacket(const uint8_t* packet, size_t length) = 0;
141 const uint8_t* packet, 139
142 size_t length) = 0; 140 // Logs an outgoing RTCP packet.
141 virtual void LogOutgoingRtcpPacket(const uint8_t* packet, size_t length) = 0;
143 142
144 // Logs an audio playout event. 143 // Logs an audio playout event.
145 virtual void LogAudioPlayout(uint32_t ssrc) = 0; 144 virtual void LogAudioPlayout(uint32_t ssrc) = 0;
146 145
147 // Logs a bitrate update from the bandwidth estimator based on packet loss. 146 // Logs a bitrate update from the bandwidth estimator based on packet loss.
148 virtual void LogLossBasedBweUpdate(int32_t bitrate_bps, 147 virtual void LogLossBasedBweUpdate(int32_t bitrate_bps,
149 uint8_t fraction_loss, 148 uint8_t fraction_loss,
150 int32_t total_packets) = 0; 149 int32_t total_packets) = 0;
151 150
152 // Logs a bitrate update from the bandwidth estimator based on delay changes. 151 // Logs a bitrate update from the bandwidth estimator based on delay changes.
(...skipping 39 matching lines...) Expand 10 before | Expand all | Expand 10 after
192 int64_t max_size_bytes) override { 191 int64_t max_size_bytes) override {
193 return false; 192 return false;
194 } 193 }
195 void StopLogging() override {} 194 void StopLogging() override {}
196 void LogVideoReceiveStreamConfig( 195 void LogVideoReceiveStreamConfig(
197 const rtclog::StreamConfig& config) override {} 196 const rtclog::StreamConfig& config) override {}
198 void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {} 197 void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {}
199 void LogAudioReceiveStreamConfig( 198 void LogAudioReceiveStreamConfig(
200 const rtclog::StreamConfig& config) override {} 199 const rtclog::StreamConfig& config) override {}
201 void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {} 200 void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {}
202 void LogRtpHeader(PacketDirection direction, 201 void LogIncomingRtpHeader(const uint8_t* header,
203 const uint8_t* header, 202 size_t packet_length) override {}
204 size_t packet_length) override {} 203 void LogOutgoingRtpHeader(const uint8_t* header,
205 void LogRtpHeader(PacketDirection direction, 204 size_t packet_length,
206 const uint8_t* header, 205 int probe_cluster_id) override {}
207 size_t packet_length, 206 void LogIncomingRtcpPacket(const uint8_t* packet, size_t length) override {}
208 int probe_cluster_id) override {} 207 void LogOutgoingRtcpPacket(const uint8_t* packet, size_t length) override {}
209 void LogRtcpPacket(PacketDirection direction,
210 const uint8_t* packet,
211 size_t length) override {}
212 void LogAudioPlayout(uint32_t ssrc) override {} 208 void LogAudioPlayout(uint32_t ssrc) override {}
213 void LogLossBasedBweUpdate(int32_t bitrate_bps, 209 void LogLossBasedBweUpdate(int32_t bitrate_bps,
214 uint8_t fraction_loss, 210 uint8_t fraction_loss,
215 int32_t total_packets) override {} 211 int32_t total_packets) override {}
216 void LogDelayBasedBweUpdate(int32_t bitrate_bps, 212 void LogDelayBasedBweUpdate(int32_t bitrate_bps,
217 BandwidthUsage detector_state) override {} 213 BandwidthUsage detector_state) override {}
218 void LogAudioNetworkAdaptation( 214 void LogAudioNetworkAdaptation(
219 const AudioEncoderRuntimeConfig& config) override {} 215 const AudioEncoderRuntimeConfig& config) override {}
220 void LogProbeClusterCreated(int id, 216 void LogProbeClusterCreated(int id,
221 int bitrate_bps, 217 int bitrate_bps,
222 int min_probes, 218 int min_probes,
223 int min_bytes) override{}; 219 int min_bytes) override{};
224 void LogProbeResultSuccess(int id, int bitrate_bps) override{}; 220 void LogProbeResultSuccess(int id, int bitrate_bps) override{};
225 void LogProbeResultFailure(int id, 221 void LogProbeResultFailure(int id,
226 ProbeFailureReason failure_reason) override{}; 222 ProbeFailureReason failure_reason) override{};
227 }; 223 };
228 224
229 } // namespace webrtc 225 } // namespace webrtc
230 226
231 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ 227 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
OLDNEW
« no previous file with comments | « webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h ('k') | webrtc/logging/rtc_event_log/rtc_event_log.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698