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Side by Side Diff: webrtc/api/webrtcsession.h

Issue 1975453002: Add PeerConnection IsClosed check. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Make it an unique_ptr when passing the TransportController to WebRtcSession. Created 4 years, 6 months ago
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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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