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Side by Side Diff: content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc

Issue 1615433002: Roll WebRTC 11523:11548, Libjingle 11522:11545 (Closed) Base URL: http://chromium.googlesource.com/chromium/src.git@master
Patch Set: Rolling to webrtc@11548 instead to pull in a fix Created 4 years, 10 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory. h" 5 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory. h"
6 6
7 #include <stddef.h> 7 #include <stddef.h>
8 8
9 #include "base/logging.h" 9 #include "base/logging.h"
10 #include "base/strings/utf_string_conversions.h" 10 #include "base/strings/utf_string_conversions.h"
11 #include "content/renderer/media/mock_peer_connection_impl.h" 11 #include "content/renderer/media/mock_peer_connection_impl.h"
12 #include "content/renderer/media/webaudio_capturer_source.h" 12 #include "content/renderer/media/webaudio_capturer_source.h"
13 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 13 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
14 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" 14 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
15 #include "content/renderer/media/webrtc_audio_capturer.h" 15 #include "content/renderer/media/webrtc_audio_capturer.h"
16 #include "content/renderer/media/webrtc_local_audio_track.h" 16 #include "content/renderer/media/webrtc_local_audio_track.h"
17 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" 17 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
18 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" 18 #include "third_party/webrtc/api/mediastreaminterface.h"
19 #include "third_party/webrtc/base/scoped_ref_ptr.h" 19 #include "third_party/webrtc/base/scoped_ref_ptr.h"
20 #include "third_party/webrtc/media/base/videocapturer.h" 20 #include "third_party/webrtc/media/base/videocapturer.h"
21 21
22 using webrtc::AudioSourceInterface; 22 using webrtc::AudioSourceInterface;
23 using webrtc::AudioTrackInterface; 23 using webrtc::AudioTrackInterface;
24 using webrtc::AudioTrackVector; 24 using webrtc::AudioTrackVector;
25 using webrtc::IceCandidateCollection; 25 using webrtc::IceCandidateCollection;
26 using webrtc::IceCandidateInterface; 26 using webrtc::IceCandidateInterface;
27 using webrtc::MediaStreamInterface; 27 using webrtc::MediaStreamInterface;
28 using webrtc::ObserverInterface; 28 using webrtc::ObserverInterface;
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533 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL, 533 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL,
534 audio_source); 534 audio_source);
535 } 535 }
536 536
537 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack( 537 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack(
538 WebRtcLocalAudioTrack* audio_track) { 538 WebRtcLocalAudioTrack* audio_track) {
539 audio_track->Start(); 539 audio_track->Start();
540 } 540 }
541 541
542 } // namespace content 542 } // namespace content
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