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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" | |
22 #include "webrtc/base/socket.h" | 21 #include "webrtc/base/socket.h" |
23 #include "webrtc/video_receive_stream.h" | 22 #include "webrtc/video_receive_stream.h" |
24 #include "webrtc/video_send_stream.h" | 23 #include "webrtc/video_send_stream.h" |
25 | 24 |
26 namespace webrtc { | 25 namespace webrtc { |
27 | 26 |
28 class AudioProcessing; | 27 class AudioProcessing; |
29 | 28 |
30 const char* Version(); | 29 const char* Version(); |
31 | 30 |
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141 // for each stream separately. Right now it's global per media type. | 140 // for each stream separately. Right now it's global per media type. |
142 virtual void SignalChannelNetworkState(MediaType media, | 141 virtual void SignalChannelNetworkState(MediaType media, |
143 NetworkState state) = 0; | 142 NetworkState state) = 0; |
144 | 143 |
145 virtual void OnNetworkRouteChanged( | 144 virtual void OnNetworkRouteChanged( |
146 const std::string& transport_name, | 145 const std::string& transport_name, |
147 const rtc::NetworkRoute& network_route) = 0; | 146 const rtc::NetworkRoute& network_route) = 0; |
148 | 147 |
149 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 148 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
150 | 149 |
151 virtual bool StartEventLog(rtc::PlatformFile log_file, | |
152 int64_t max_size_bytes) = 0; | |
153 virtual void StopEventLog() = 0; | |
154 | |
155 virtual ~Call() {} | 150 virtual ~Call() {} |
156 }; | 151 }; |
157 | 152 |
158 } // namespace webrtc | 153 } // namespace webrtc |
159 | 154 |
160 #endif // WEBRTC_CALL_H_ | 155 #endif // WEBRTC_CALL_H_ |
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