| Index: content/renderer/media/mock_media_stream_dependency_factory.cc
|
| diff --git a/content/renderer/media/mock_media_stream_dependency_factory.cc b/content/renderer/media/mock_media_stream_dependency_factory.cc
|
| index 5b58c5ee157769f69d6edea1123abbbcb55211d3..26c3275e2caa895e2f35c83eb91733e0ac069e64 100644
|
| --- a/content/renderer/media/mock_media_stream_dependency_factory.cc
|
| +++ b/content/renderer/media/mock_media_stream_dependency_factory.cc
|
| @@ -7,6 +7,7 @@
|
| #include "base/logging.h"
|
| #include "base/strings/utf_string_conversions.h"
|
| #include "content/renderer/media/mock_peer_connection_impl.h"
|
| +#include "content/renderer/media/webaudio_capturer_source.h"
|
| #include "content/renderer/media/webrtc_audio_capturer.h"
|
| #include "content/renderer/media/webrtc_local_audio_track.h"
|
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
| @@ -406,7 +407,7 @@ MockMediaStreamDependencyFactory::CreateLocalVideoSource(
|
| return last_video_source_;
|
| }
|
|
|
| -scoped_refptr<WebRtcAudioCapturer>
|
| +scoped_refptr<WebAudioCapturerSource>
|
| MockMediaStreamDependencyFactory::CreateWebAudioSource(
|
| WebKit::WebMediaStreamSource* source,
|
| RTCMediaConstraints* constraints) {
|
| @@ -448,12 +449,14 @@ scoped_refptr<webrtc::AudioTrackInterface>
|
| MockMediaStreamDependencyFactory::CreateLocalAudioTrack(
|
| const std::string& id,
|
| const scoped_refptr<WebRtcAudioCapturer>& capturer,
|
| + WebAudioCapturerSource* webaudio_source,
|
| webrtc::AudioSourceInterface* source,
|
| const webrtc::MediaConstraintsInterface* constraints) {
|
| DCHECK(mock_pc_factory_created_);
|
| DCHECK(!capturer.get());
|
| return WebRtcLocalAudioTrack::Create(
|
| - id, WebRtcAudioCapturer::CreateCapturer(), source, constraints);
|
| + id, WebRtcAudioCapturer::CreateCapturer(), webaudio_source,
|
| + source, constraints);
|
| }
|
|
|
| SessionDescriptionInterface*
|
|
|