| Index: remoting/protocol/webrtc_transport.cc
|
| diff --git a/remoting/protocol/webrtc_transport.cc b/remoting/protocol/webrtc_transport.cc
|
| index e13544ecadb0a600a7cb45ef8b64cee3f99d9473..b3f759ac652728238ec39569e5ad9ce69f814dce 100644
|
| --- a/remoting/protocol/webrtc_transport.cc
|
| +++ b/remoting/protocol/webrtc_transport.cc
|
| @@ -24,10 +24,10 @@
|
| #include "remoting/protocol/port_allocator_factory.h"
|
| #include "remoting/protocol/stream_message_pipe_adapter.h"
|
| #include "remoting/protocol/transport_context.h"
|
| +#include "remoting/protocol/webrtc_audio_module.h"
|
| #include "remoting/protocol/webrtc_dummy_video_encoder.h"
|
| #include "third_party/webrtc/api/test/fakeconstraints.h"
|
| #include "third_party/webrtc/libjingle/xmllite/xmlelement.h"
|
| -#include "third_party/webrtc/modules/audio_device/include/fake_audio_device.h"
|
|
|
| using buzz::QName;
|
| using buzz::XmlElement;
|
| @@ -146,8 +146,11 @@ class WebrtcTransport::PeerConnectionWrapper
|
| std::unique_ptr<cricket::PortAllocator> port_allocator,
|
| base::WeakPtr<WebrtcTransport> transport)
|
| : transport_(transport) {
|
| + scoped_refptr<WebrtcAudioModule> audio_module =
|
| + new rtc::RefCountedObject<WebrtcAudioModule>();
|
| +
|
| peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
|
| - worker_thread, rtc::Thread::Current(), &fake_audio_device_module_,
|
| + worker_thread, rtc::Thread::Current(), audio_module.get(),
|
| encoder_factory.release(), nullptr);
|
|
|
| webrtc::FakeConstraints constraints;
|
| @@ -208,7 +211,7 @@ class WebrtcTransport::PeerConnectionWrapper
|
| }
|
|
|
| private:
|
| - webrtc::FakeAudioDeviceModule fake_audio_device_module_;
|
| + scoped_refptr<WebrtcAudioModule> audio_module_;
|
| scoped_refptr<webrtc::PeerConnectionFactoryInterface>
|
| peer_connection_factory_;
|
| scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
|
|
|