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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" | 17 #include "webrtc/api/call/audio_receive_stream.h" |
| 18 #include "webrtc/api/call/audio_send_stream.h" | 18 #include "webrtc/api/call/audio_send_stream.h" |
| 19 #include "webrtc/base/platform_file.h" | 19 #include "webrtc/base/platform_file.h" |
| 20 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_networ
k_adaptor.h" |
| 20 #include "webrtc/video_receive_stream.h" | 21 #include "webrtc/video_receive_stream.h" |
| 21 #include "webrtc/video_send_stream.h" | 22 #include "webrtc/video_send_stream.h" |
| 22 | 23 |
| 23 namespace webrtc { | 24 namespace webrtc { |
| 24 | 25 |
| 25 // Forward declaration of storage class that is automatically generated from | 26 // Forward declaration of storage class that is automatically generated from |
| 26 // the protobuf file. | 27 // the protobuf file. |
| 27 namespace rtclog { | 28 namespace rtclog { |
| 28 class EventStream; | 29 class EventStream; |
| 29 } // namespace rtclog | 30 } // namespace rtclog |
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| 101 size_t length) = 0; | 102 size_t length) = 0; |
| 102 | 103 |
| 103 // Logs an audio playout event. | 104 // Logs an audio playout event. |
| 104 virtual void LogAudioPlayout(uint32_t ssrc) = 0; | 105 virtual void LogAudioPlayout(uint32_t ssrc) = 0; |
| 105 | 106 |
| 106 // Logs a bitrate update from the bandwidth estimator based on packet loss. | 107 // Logs a bitrate update from the bandwidth estimator based on packet loss. |
| 107 virtual void LogBwePacketLossEvent(int32_t bitrate, | 108 virtual void LogBwePacketLossEvent(int32_t bitrate, |
| 108 uint8_t fraction_loss, | 109 uint8_t fraction_loss, |
| 109 int32_t total_packets) = 0; | 110 int32_t total_packets) = 0; |
| 110 | 111 |
| 112 // Logs audio encoder configuration changes. |
| 113 virtual void LogAudioEncoderConfig( |
| 114 const AudioNetworkAdaptor::EncoderRuntimeConfig& config) = 0; |
| 115 |
| 111 // Reads an RtcEventLog file and returns true when reading was successful. | 116 // Reads an RtcEventLog file and returns true when reading was successful. |
| 112 // The result is stored in the given EventStream object. | 117 // The result is stored in the given EventStream object. |
| 113 // The order of the events in the EventStream is implementation defined. | 118 // The order of the events in the EventStream is implementation defined. |
| 114 // The current implementation writes a LOG_START event, then the old | 119 // The current implementation writes a LOG_START event, then the old |
| 115 // configurations, then the remaining events in timestamp order and finally | 120 // configurations, then the remaining events in timestamp order and finally |
| 116 // a LOG_END event. However, this might change without further notice. | 121 // a LOG_END event. However, this might change without further notice. |
| 117 // TODO(terelius): Change result type to a vector? | 122 // TODO(terelius): Change result type to a vector? |
| 118 static bool ParseRtcEventLog(const std::string& file_name, | 123 static bool ParseRtcEventLog(const std::string& file_name, |
| 119 rtclog::EventStream* result); | 124 rtclog::EventStream* result); |
| 120 }; | 125 }; |
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| 142 const uint8_t* header, | 147 const uint8_t* header, |
| 143 size_t packet_length) override {} | 148 size_t packet_length) override {} |
| 144 void LogRtcpPacket(PacketDirection direction, | 149 void LogRtcpPacket(PacketDirection direction, |
| 145 MediaType media_type, | 150 MediaType media_type, |
| 146 const uint8_t* packet, | 151 const uint8_t* packet, |
| 147 size_t length) override {} | 152 size_t length) override {} |
| 148 void LogAudioPlayout(uint32_t ssrc) override {} | 153 void LogAudioPlayout(uint32_t ssrc) override {} |
| 149 void LogBwePacketLossEvent(int32_t bitrate, | 154 void LogBwePacketLossEvent(int32_t bitrate, |
| 150 uint8_t fraction_loss, | 155 uint8_t fraction_loss, |
| 151 int32_t total_packets) override {} | 156 int32_t total_packets) override {} |
| 157 void LogAudioEncoderConfig( |
| 158 const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override{}; |
| 152 }; | 159 }; |
| 153 | 160 |
| 154 } // namespace webrtc | 161 } // namespace webrtc |
| 155 | 162 |
| 156 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ | 163 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
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