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Side by Side Diff: talk/app/webrtc/objc/RTCPeerConnectionFactory.mm

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 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,
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39 #import "RTCMediaSource+Internal.h" 39 #import "RTCMediaSource+Internal.h"
40 #import "RTCMediaStream+Internal.h" 40 #import "RTCMediaStream+Internal.h"
41 #import "RTCMediaStreamTrack+Internal.h" 41 #import "RTCMediaStreamTrack+Internal.h"
42 #import "RTCPeerConnection+Internal.h" 42 #import "RTCPeerConnection+Internal.h"
43 #import "RTCPeerConnectionDelegate.h" 43 #import "RTCPeerConnectionDelegate.h"
44 #import "RTCPeerConnectionInterface+Internal.h" 44 #import "RTCPeerConnectionInterface+Internal.h"
45 #import "RTCVideoCapturer+Internal.h" 45 #import "RTCVideoCapturer+Internal.h"
46 #import "RTCVideoSource+Internal.h" 46 #import "RTCVideoSource+Internal.h"
47 #import "RTCVideoTrack+Internal.h" 47 #import "RTCVideoTrack+Internal.h"
48 48
49 #include "talk/app/webrtc/audiotrack.h" 49 #include "webrtc/api/audiotrack.h"
50 #include "talk/app/webrtc/mediastreaminterface.h" 50 #include "webrtc/api/mediastreaminterface.h"
51 #include "talk/app/webrtc/peerconnectioninterface.h" 51 #include "webrtc/api/peerconnectioninterface.h"
52 #include "talk/app/webrtc/videosourceinterface.h" 52 #include "webrtc/api/videosourceinterface.h"
53 #include "talk/app/webrtc/videotrack.h" 53 #include "webrtc/api/videotrack.h"
54 #include "webrtc/base/logging.h" 54 #include "webrtc/base/logging.h"
55 #include "webrtc/base/ssladapter.h" 55 #include "webrtc/base/ssladapter.h"
56 56
57 @implementation RTCPeerConnectionFactory { 57 @implementation RTCPeerConnectionFactory {
58 rtc::scoped_ptr<rtc::Thread> _signalingThread; 58 rtc::scoped_ptr<rtc::Thread> _signalingThread;
59 rtc::scoped_ptr<rtc::Thread> _workerThread; 59 rtc::scoped_ptr<rtc::Thread> _workerThread;
60 } 60 }
61 61
62 @synthesize nativeFactory = _nativeFactory; 62 @synthesize nativeFactory = _nativeFactory;
63 63
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139 return [[RTCVideoTrack alloc] initWithMediaTrack:track]; 139 return [[RTCVideoTrack alloc] initWithMediaTrack:track];
140 } 140 }
141 141
142 - (RTCAudioTrack*)audioTrackWithID:(NSString*)audioId { 142 - (RTCAudioTrack*)audioTrackWithID:(NSString*)audioId {
143 rtc::scoped_refptr<webrtc::AudioTrackInterface> track = 143 rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
144 self.nativeFactory->CreateAudioTrack([audioId UTF8String], NULL); 144 self.nativeFactory->CreateAudioTrack([audioId UTF8String], NULL);
145 return [[RTCAudioTrack alloc] initWithMediaTrack:track]; 145 return [[RTCAudioTrack alloc] initWithMediaTrack:track];
146 } 146 }
147 147
148 @end 148 @end
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