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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 |
11 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ | 11 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
12 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ | 12 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
| 18 #include "webrtc/api/array_view.h" |
18 // TODO(eladalon): Get rid of this later in the CL-stack. | 19 // TODO(eladalon): Get rid of this later in the CL-stack. |
19 #include "webrtc/api/rtpparameters.h" | 20 #include "webrtc/api/rtpparameters.h" |
20 #include "webrtc/common_types.h" | 21 #include "webrtc/common_types.h" |
21 #include "webrtc/rtc_base/platform_file.h" | 22 #include "webrtc/rtc_base/platform_file.h" |
22 | 23 |
23 namespace webrtc { | 24 namespace webrtc { |
24 | 25 |
25 namespace rtclog { | 26 namespace rtclog { |
26 class EventStream; // Storage class automatically generated from protobuf. | 27 class EventStream; // Storage class automatically generated from protobuf. |
27 // TODO(eladalon): Get rid of this when deprecated methods are removed. | 28 // TODO(eladalon): Get rid of this when deprecated methods are removed. |
28 struct StreamConfig; | 29 struct StreamConfig; |
29 } // namespace rtclog | 30 } // namespace rtclog |
30 | 31 |
31 class Clock; | 32 class Clock; |
32 struct AudioEncoderRuntimeConfig; | 33 struct AudioEncoderRuntimeConfig; |
| 34 class RtpPacketReceived; |
| 35 class RtpPacketToSend; |
33 | 36 |
34 enum class MediaType; | 37 enum class MediaType; |
35 enum class BandwidthUsage; | 38 enum class BandwidthUsage; |
36 | 39 |
37 enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket }; | 40 enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket }; |
38 enum ProbeFailureReason { | 41 enum ProbeFailureReason { |
39 kInvalidSendReceiveInterval, | 42 kInvalidSendReceiveInterval, |
40 kInvalidSendReceiveRatio, | 43 kInvalidSendReceiveRatio, |
41 kTimeout | 44 kTimeout |
42 }; | 45 }; |
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91 // Logs configuration information for a video send stream. | 94 // Logs configuration information for a video send stream. |
92 virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0; | 95 virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0; |
93 | 96 |
94 // Logs configuration information for an audio receive stream. | 97 // Logs configuration information for an audio receive stream. |
95 virtual void LogAudioReceiveStreamConfig( | 98 virtual void LogAudioReceiveStreamConfig( |
96 const rtclog::StreamConfig& config) = 0; | 99 const rtclog::StreamConfig& config) = 0; |
97 | 100 |
98 // Logs configuration information for an audio send stream. | 101 // Logs configuration information for an audio send stream. |
99 virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) = 0; | 102 virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) = 0; |
100 | 103 |
101 // Logs the header of an incoming or outgoing RTP packet. packet_length | 104 RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction, |
| 105 const uint8_t* header, |
| 106 size_t packet_length) {} |
| 107 |
| 108 RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction, |
| 109 const uint8_t* header, |
| 110 size_t packet_length, |
| 111 int probe_cluster_id) {} |
| 112 |
| 113 // Logs the header of an incoming RTP packet. |packet_length| |
102 // is the total length of the packet, including both header and payload. | 114 // is the total length of the packet, including both header and payload. |
103 virtual void LogRtpHeader(PacketDirection direction, | 115 virtual void LogIncomingRtpHeader(const RtpPacketReceived& packet) = 0; |
104 const uint8_t* header, | |
105 size_t packet_length) = 0; | |
106 | 116 |
107 // Same as above but used on the sender side to log packets that are part of | 117 // Logs the header of an incoming RTP packet. |packet_length| |
108 // a probe cluster. | 118 // is the total length of the packet, including both header and payload. |
109 virtual void LogRtpHeader(PacketDirection direction, | 119 virtual void LogOutgoingRtpHeader(const RtpPacketToSend& packet, |
110 const uint8_t* header, | 120 int probe_cluster_id) = 0; |
111 size_t packet_length, | |
112 int probe_cluster_id) = 0; | |
113 | 121 |
114 // Logs an incoming or outgoing RTCP packet. | 122 RTC_DEPRECATED virtual void LogRtcpPacket(PacketDirection direction, |
115 virtual void LogRtcpPacket(PacketDirection direction, | 123 const uint8_t* header, |
116 const uint8_t* packet, | 124 size_t packet_length) {} |
117 size_t length) = 0; | 125 |
| 126 // Logs an incoming RTCP packet. |
| 127 virtual void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0; |
| 128 |
| 129 // Logs an outgoing RTCP packet. |
| 130 virtual void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0; |
118 | 131 |
119 // Logs an audio playout event. | 132 // Logs an audio playout event. |
120 virtual void LogAudioPlayout(uint32_t ssrc) = 0; | 133 virtual void LogAudioPlayout(uint32_t ssrc) = 0; |
121 | 134 |
122 // Logs a bitrate update from the bandwidth estimator based on packet loss. | 135 // Logs a bitrate update from the bandwidth estimator based on packet loss. |
123 virtual void LogLossBasedBweUpdate(int32_t bitrate_bps, | 136 virtual void LogLossBasedBweUpdate(int32_t bitrate_bps, |
124 uint8_t fraction_loss, | 137 uint8_t fraction_loss, |
125 int32_t total_packets) = 0; | 138 int32_t total_packets) = 0; |
126 | 139 |
127 // Logs a bitrate update from the bandwidth estimator based on delay changes. | 140 // Logs a bitrate update from the bandwidth estimator based on delay changes. |
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157 int64_t max_size_bytes) override { | 170 int64_t max_size_bytes) override { |
158 return false; | 171 return false; |
159 } | 172 } |
160 void StopLogging() override {} | 173 void StopLogging() override {} |
161 void LogVideoReceiveStreamConfig( | 174 void LogVideoReceiveStreamConfig( |
162 const rtclog::StreamConfig& config) override {} | 175 const rtclog::StreamConfig& config) override {} |
163 void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {} | 176 void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {} |
164 void LogAudioReceiveStreamConfig( | 177 void LogAudioReceiveStreamConfig( |
165 const rtclog::StreamConfig& config) override {} | 178 const rtclog::StreamConfig& config) override {} |
166 void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {} | 179 void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {} |
167 void LogRtpHeader(PacketDirection direction, | 180 void LogIncomingRtpHeader(const RtpPacketReceived& packet) override {} |
168 const uint8_t* header, | 181 void LogOutgoingRtpHeader(const RtpPacketToSend& packet, |
169 size_t packet_length) override {} | 182 int probe_cluster_id) override {} |
170 void LogRtpHeader(PacketDirection direction, | 183 void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {} |
171 const uint8_t* header, | 184 void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {} |
172 size_t packet_length, | |
173 int probe_cluster_id) override {} | |
174 void LogRtcpPacket(PacketDirection direction, | |
175 const uint8_t* packet, | |
176 size_t length) override {} | |
177 void LogAudioPlayout(uint32_t ssrc) override {} | 185 void LogAudioPlayout(uint32_t ssrc) override {} |
178 void LogLossBasedBweUpdate(int32_t bitrate_bps, | 186 void LogLossBasedBweUpdate(int32_t bitrate_bps, |
179 uint8_t fraction_loss, | 187 uint8_t fraction_loss, |
180 int32_t total_packets) override {} | 188 int32_t total_packets) override {} |
181 void LogDelayBasedBweUpdate(int32_t bitrate_bps, | 189 void LogDelayBasedBweUpdate(int32_t bitrate_bps, |
182 BandwidthUsage detector_state) override {} | 190 BandwidthUsage detector_state) override {} |
183 void LogAudioNetworkAdaptation( | 191 void LogAudioNetworkAdaptation( |
184 const AudioEncoderRuntimeConfig& config) override {} | 192 const AudioEncoderRuntimeConfig& config) override {} |
185 void LogProbeClusterCreated(int id, | 193 void LogProbeClusterCreated(int id, |
186 int bitrate_bps, | 194 int bitrate_bps, |
187 int min_probes, | 195 int min_probes, |
188 int min_bytes) override{}; | 196 int min_bytes) override{}; |
189 void LogProbeResultSuccess(int id, int bitrate_bps) override{}; | 197 void LogProbeResultSuccess(int id, int bitrate_bps) override{}; |
190 void LogProbeResultFailure(int id, | 198 void LogProbeResultFailure(int id, |
191 ProbeFailureReason failure_reason) override{}; | 199 ProbeFailureReason failure_reason) override{}; |
192 }; | 200 }; |
193 | 201 |
194 } // namespace webrtc | 202 } // namespace webrtc |
195 | 203 |
196 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ | 204 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
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