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

Issue 1547073003: Switch to standard integer types in content/renderer/. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 12 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>
8
7 #include "base/logging.h" 9 #include "base/logging.h"
8 #include "base/strings/utf_string_conversions.h" 10 #include "base/strings/utf_string_conversions.h"
9 #include "content/renderer/media/mock_peer_connection_impl.h" 11 #include "content/renderer/media/mock_peer_connection_impl.h"
10 #include "content/renderer/media/webaudio_capturer_source.h" 12 #include "content/renderer/media/webaudio_capturer_source.h"
11 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 13 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
12 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" 14 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
13 #include "content/renderer/media/webrtc_audio_capturer.h" 15 #include "content/renderer/media/webrtc_audio_capturer.h"
14 #include "content/renderer/media/webrtc_local_audio_track.h" 16 #include "content/renderer/media/webrtc_local_audio_track.h"
15 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" 17 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
16 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" 18 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
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537 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL, 539 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL,
538 audio_source); 540 audio_source);
539 } 541 }
540 542
541 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack( 543 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack(
542 WebRtcLocalAudioTrack* audio_track) { 544 WebRtcLocalAudioTrack* audio_track) {
543 audio_track->Start(); 545 audio_track->Start();
544 } 546 }
545 547
546 } // namespace content 548 } // namespace content
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