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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 | 16 |
| 17 #include "webrtc/api/call/audio_receive_stream.h" |
| 18 #include "webrtc/api/call/audio_send_stream.h" |
17 #include "webrtc/base/platform_file.h" | 19 #include "webrtc/base/platform_file.h" |
18 #include "webrtc/video_receive_stream.h" | 20 #include "webrtc/video_receive_stream.h" |
19 #include "webrtc/video_send_stream.h" | 21 #include "webrtc/video_send_stream.h" |
20 | 22 |
21 namespace webrtc { | 23 namespace webrtc { |
22 | 24 |
23 // Forward declaration of storage class that is automatically generated from | 25 // Forward declaration of storage class that is automatically generated from |
24 // the protobuf file. | 26 // the protobuf file. |
25 namespace rtclog { | 27 namespace rtclog { |
26 class EventStream; | 28 class EventStream; |
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70 virtual void StopLogging() = 0; | 72 virtual void StopLogging() = 0; |
71 | 73 |
72 // Logs configuration information for webrtc::VideoReceiveStream. | 74 // Logs configuration information for webrtc::VideoReceiveStream. |
73 virtual void LogVideoReceiveStreamConfig( | 75 virtual void LogVideoReceiveStreamConfig( |
74 const webrtc::VideoReceiveStream::Config& config) = 0; | 76 const webrtc::VideoReceiveStream::Config& config) = 0; |
75 | 77 |
76 // Logs configuration information for webrtc::VideoSendStream. | 78 // Logs configuration information for webrtc::VideoSendStream. |
77 virtual void LogVideoSendStreamConfig( | 79 virtual void LogVideoSendStreamConfig( |
78 const webrtc::VideoSendStream::Config& config) = 0; | 80 const webrtc::VideoSendStream::Config& config) = 0; |
79 | 81 |
| 82 // Logs configuration information for webrtc::AudioReceiveStream. |
| 83 virtual void LogAudioReceiveStreamConfig( |
| 84 const webrtc::AudioReceiveStream::Config& config) = 0; |
| 85 |
| 86 // Logs configuration information for webrtc::AudioSendStream. |
| 87 virtual void LogAudioSendStreamConfig( |
| 88 const webrtc::AudioSendStream::Config& config) = 0; |
| 89 |
80 // Logs the header of an incoming or outgoing RTP packet. packet_length | 90 // Logs the header of an incoming or outgoing RTP packet. packet_length |
81 // is the total length of the packet, including both header and payload. | 91 // is the total length of the packet, including both header and payload. |
82 virtual void LogRtpHeader(PacketDirection direction, | 92 virtual void LogRtpHeader(PacketDirection direction, |
83 MediaType media_type, | 93 MediaType media_type, |
84 const uint8_t* header, | 94 const uint8_t* header, |
85 size_t packet_length) = 0; | 95 size_t packet_length) = 0; |
86 | 96 |
87 // Logs an incoming or outgoing RTCP packet. | 97 // Logs an incoming or outgoing RTCP packet. |
88 virtual void LogRtcpPacket(PacketDirection direction, | 98 virtual void LogRtcpPacket(PacketDirection direction, |
89 MediaType media_type, | 99 MediaType media_type, |
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116 int64_t max_size_bytes) override { | 126 int64_t max_size_bytes) override { |
117 return false; | 127 return false; |
118 } | 128 } |
119 bool StartLogging(rtc::PlatformFile platform_file, | 129 bool StartLogging(rtc::PlatformFile platform_file, |
120 int64_t max_size_bytes) override; | 130 int64_t max_size_bytes) override; |
121 void StopLogging() override {} | 131 void StopLogging() override {} |
122 void LogVideoReceiveStreamConfig( | 132 void LogVideoReceiveStreamConfig( |
123 const VideoReceiveStream::Config& config) override {} | 133 const VideoReceiveStream::Config& config) override {} |
124 void LogVideoSendStreamConfig( | 134 void LogVideoSendStreamConfig( |
125 const VideoSendStream::Config& config) override {} | 135 const VideoSendStream::Config& config) override {} |
| 136 void LogAudioReceiveStreamConfig( |
| 137 const AudioReceiveStream::Config& config) override {} |
| 138 void LogAudioSendStreamConfig( |
| 139 const AudioSendStream::Config& config) override {} |
126 void LogRtpHeader(PacketDirection direction, | 140 void LogRtpHeader(PacketDirection direction, |
127 MediaType media_type, | 141 MediaType media_type, |
128 const uint8_t* header, | 142 const uint8_t* header, |
129 size_t packet_length) override {} | 143 size_t packet_length) override {} |
130 void LogRtcpPacket(PacketDirection direction, | 144 void LogRtcpPacket(PacketDirection direction, |
131 MediaType media_type, | 145 MediaType media_type, |
132 const uint8_t* packet, | 146 const uint8_t* packet, |
133 size_t length) override {} | 147 size_t length) override {} |
134 void LogAudioPlayout(uint32_t ssrc) override {} | 148 void LogAudioPlayout(uint32_t ssrc) override {} |
135 void LogBwePacketLossEvent(int32_t bitrate, | 149 void LogBwePacketLossEvent(int32_t bitrate, |
136 uint8_t fraction_loss, | 150 uint8_t fraction_loss, |
137 int32_t total_packets) override {} | 151 int32_t total_packets) override {} |
138 }; | 152 }; |
139 | 153 |
140 } // namespace webrtc | 154 } // namespace webrtc |
141 | 155 |
142 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ | 156 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
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