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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #ifndef WEBRTC_CALL_H_ | 10 #ifndef WEBRTC_CALL_H_ |
11 #define WEBRTC_CALL_H_ | 11 #define WEBRTC_CALL_H_ |
12 | 12 |
13 #include <string> | 13 #include <string> |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
16 #include "webrtc/common_types.h" | 16 #include "webrtc/common_types.h" |
17 #include "webrtc/audio_receive_stream.h" | 17 #include "webrtc/audio_receive_stream.h" |
18 #include "webrtc/audio_send_stream.h" | 18 #include "webrtc/audio_send_stream.h" |
19 #include "webrtc/audio_state.h" | 19 #include "webrtc/audio_state.h" |
20 #include "webrtc/base/networkroute.h" | 20 #include "webrtc/base/networkroute.h" |
| 21 #include "webrtc/base/platform_file.h" |
21 #include "webrtc/base/socket.h" | 22 #include "webrtc/base/socket.h" |
22 #include "webrtc/video_receive_stream.h" | 23 #include "webrtc/video_receive_stream.h" |
23 #include "webrtc/video_send_stream.h" | 24 #include "webrtc/video_send_stream.h" |
24 | 25 |
25 namespace webrtc { | 26 namespace webrtc { |
26 | 27 |
27 class AudioProcessing; | 28 class AudioProcessing; |
28 | 29 |
29 const char* Version(); | 30 const char* Version(); |
30 | 31 |
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140 // for each stream separately. Right now it's global per media type. | 141 // for each stream separately. Right now it's global per media type. |
141 virtual void SignalChannelNetworkState(MediaType media, | 142 virtual void SignalChannelNetworkState(MediaType media, |
142 NetworkState state) = 0; | 143 NetworkState state) = 0; |
143 | 144 |
144 virtual void OnNetworkRouteChanged( | 145 virtual void OnNetworkRouteChanged( |
145 const std::string& transport_name, | 146 const std::string& transport_name, |
146 const rtc::NetworkRoute& network_route) = 0; | 147 const rtc::NetworkRoute& network_route) = 0; |
147 | 148 |
148 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 149 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
149 | 150 |
| 151 virtual bool StartEventLog(rtc::PlatformFile log_file, |
| 152 int64_t max_size_bytes) = 0; |
| 153 virtual void StopEventLog() = 0; |
| 154 |
150 virtual ~Call() {} | 155 virtual ~Call() {} |
151 }; | 156 }; |
152 | 157 |
153 } // namespace webrtc | 158 } // namespace webrtc |
154 | 159 |
155 #endif // WEBRTC_CALL_H_ | 160 #endif // WEBRTC_CALL_H_ |
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