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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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