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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 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 |
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130 STATE_INPROGRESS, // Offer/answer exchange completed. | 130 STATE_INPROGRESS, // Offer/answer exchange completed. |
131 STATE_CLOSED, // Close() was called. | 131 STATE_CLOSED, // Close() was called. |
132 }; | 132 }; |
133 | 133 |
134 enum Error { | 134 enum Error { |
135 ERROR_NONE = 0, // no error | 135 ERROR_NONE = 0, // no error |
136 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent | 136 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent |
137 ERROR_TRANSPORT = 2, // transport error of some kind | 137 ERROR_TRANSPORT = 2, // transport error of some kind |
138 }; | 138 }; |
139 | 139 |
140 WebRtcSession(webrtc::MediaControllerInterface* media_controller, | 140 WebRtcSession( |
141 rtc::Thread* network_thread, | 141 webrtc::MediaControllerInterface* media_controller, |
142 rtc::Thread* worker_thread, | 142 rtc::Thread* network_thread, |
143 rtc::Thread* signaling_thread, | 143 rtc::Thread* worker_thread, |
144 cricket::PortAllocator* port_allocator); | 144 rtc::Thread* signaling_thread, |
| 145 cricket::PortAllocator* port_allocator, |
| 146 std::unique_ptr<cricket::TransportController> transport_controller); |
145 virtual ~WebRtcSession(); | 147 virtual ~WebRtcSession(); |
146 | 148 |
147 // These are const to allow them to be called from const methods. | 149 // These are const to allow them to be called from const methods. |
148 rtc::Thread* worker_thread() const { return worker_thread_; } | 150 rtc::Thread* worker_thread() const { return worker_thread_; } |
149 rtc::Thread* signaling_thread() const { return signaling_thread_; } | 151 rtc::Thread* signaling_thread() const { return signaling_thread_; } |
150 | 152 |
151 // The ID of this session. | 153 // The ID of this session. |
152 const std::string& id() const { return sid_; } | 154 const std::string& id() const { return sid_; } |
153 | 155 |
154 bool Initialize( | 156 bool Initialize( |
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530 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; | 532 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; |
531 | 533 |
532 bool received_first_video_packet_ = false; | 534 bool received_first_video_packet_ = false; |
533 bool received_first_audio_packet_ = false; | 535 bool received_first_audio_packet_ = false; |
534 | 536 |
535 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); | 537 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); |
536 }; | 538 }; |
537 } // namespace webrtc | 539 } // namespace webrtc |
538 | 540 |
539 #endif // WEBRTC_API_WEBRTCSESSION_H_ | 541 #endif // WEBRTC_API_WEBRTCSESSION_H_ |
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