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Side by Side Diff: talk/app/webrtc/test/peerconnectiontestwrapper.h

Issue 1236023010: In PeerConnectionTestWrapper, put audio input on a separate thread. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: resolving patch conflicts due to splitting this CL Created 5 years, 4 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2013 Google Inc. 3 * Copyright 2013 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 15 matching lines...) Expand all
26 */ 26 */
27 27
28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
30 30
31 #include "talk/app/webrtc/peerconnectioninterface.h" 31 #include "talk/app/webrtc/peerconnectioninterface.h"
32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" 32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
33 #include "talk/app/webrtc/test/fakeconstraints.h" 33 #include "talk/app/webrtc/test/fakeconstraints.h"
34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" 34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
35 #include "webrtc/base/sigslot.h" 35 #include "webrtc/base/sigslot.h"
36 #include "webrtc/base/thread.h"
37 36
38 namespace webrtc { 37 namespace webrtc {
39 class DtlsIdentityStoreInterface; 38 class DtlsIdentityStoreInterface;
40 class PortAllocatorFactoryInterface; 39 class PortAllocatorFactoryInterface;
41 } 40 }
42 41
43 class PeerConnectionTestWrapper 42 class PeerConnectionTestWrapper
44 : public webrtc::PeerConnectionObserver, 43 : public webrtc::PeerConnectionObserver,
45 public webrtc::CreateSessionDescriptionObserver, 44 public webrtc::CreateSessionDescriptionObserver,
46 public sigslot::has_slots<> { 45 public sigslot::has_slots<> {
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114 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> 113 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
115 allocator_factory_; 114 allocator_factory_;
116 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; 115 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
117 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> 116 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
118 peer_connection_factory_; 117 peer_connection_factory_;
119 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; 118 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
120 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; 119 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
121 }; 120 };
122 121
123 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 122 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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