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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 2004 Google Inc. 3 * Copyright 2004 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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26 */ 26 */
27 27
28 #include "talk/session/media/channelmanager.h" 28 #include "talk/session/media/channelmanager.h"
29 29
30 #ifdef HAVE_CONFIG_H 30 #ifdef HAVE_CONFIG_H
31 #include <config.h> 31 #include <config.h>
32 #endif 32 #endif
33 33
34 #include <algorithm> 34 #include <algorithm>
35 35
36 #include "talk/app/webrtc/mediacontroller.h"
37 #include "webrtc/media/base/capturemanager.h"
38 #include "webrtc/media/base/device.h"
39 #include "webrtc/media/base/hybriddataengine.h"
40 #include "webrtc/media/base/rtpdataengine.h"
41 #include "webrtc/media/base/videocapturer.h"
42 #ifdef HAVE_SCTP
43 #include "webrtc/media/sctp/sctpdataengine.h"
44 #endif
45 #include "talk/session/media/srtpfilter.h" 36 #include "talk/session/media/srtpfilter.h"
37 #include "webrtc/api/mediacontroller.h"
46 #include "webrtc/base/bind.h" 38 #include "webrtc/base/bind.h"
47 #include "webrtc/base/common.h" 39 #include "webrtc/base/common.h"
48 #include "webrtc/base/logging.h" 40 #include "webrtc/base/logging.h"
49 #include "webrtc/base/sigslotrepeater.h" 41 #include "webrtc/base/sigslotrepeater.h"
50 #include "webrtc/base/stringencode.h" 42 #include "webrtc/base/stringencode.h"
51 #include "webrtc/base/stringutils.h" 43 #include "webrtc/base/stringutils.h"
52 #include "webrtc/base/trace_event.h" 44 #include "webrtc/base/trace_event.h"
45 #include "webrtc/media/base/capturemanager.h"
46 #include "webrtc/media/base/device.h"
47 #include "webrtc/media/base/hybriddataengine.h"
48 #include "webrtc/media/base/rtpdataengine.h"
49 #include "webrtc/media/base/videocapturer.h"
50 #ifdef HAVE_SCTP
51 #include "webrtc/media/sctp/sctpdataengine.h"
52 #endif
53 53
54 namespace cricket { 54 namespace cricket {
55 55
56 enum { 56 enum {
57 MSG_VIDEOCAPTURESTATE = 1, 57 MSG_VIDEOCAPTURESTATE = 1,
58 }; 58 };
59 59
60 using rtc::Bind; 60 using rtc::Bind;
61 61
62 static const int kNotSetOutputVolume = -1; 62 static const int kNotSetOutputVolume = -1;
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568 return worker_thread_->Invoke<bool>( 568 return worker_thread_->Invoke<bool>(
569 Bind(&MediaEngineInterface::StartRtcEventLog, media_engine_.get(), file)); 569 Bind(&MediaEngineInterface::StartRtcEventLog, media_engine_.get(), file));
570 } 570 }
571 571
572 void ChannelManager::StopRtcEventLog() { 572 void ChannelManager::StopRtcEventLog() {
573 worker_thread_->Invoke<void>( 573 worker_thread_->Invoke<void>(
574 Bind(&MediaEngineInterface::StopRtcEventLog, media_engine_.get())); 574 Bind(&MediaEngineInterface::StopRtcEventLog, media_engine_.get()));
575 } 575 }
576 576
577 } // namespace cricket 577 } // namespace cricket
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