| // Copyright 2016 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "remoting/protocol/webrtc_audio_stream.h" |
| |
| #include "base/location.h" |
| #include "base/logging.h" |
| #include "base/single_thread_task_runner.h" |
| #include "remoting/base/constants.h" |
| #include "remoting/protocol/audio_source.h" |
| #include "remoting/protocol/webrtc_audio_source_adapter.h" |
| #include "remoting/protocol/webrtc_transport.h" |
| #include "third_party/webrtc/api/mediastreaminterface.h" |
| #include "third_party/webrtc/api/peerconnectioninterface.h" |
| #include "third_party/webrtc/rtc_base/refcount.h" |
| |
| namespace remoting { |
| namespace protocol { |
| |
| const char kAudioStreamLabel[] = "audio_stream"; |
| const char kAudioTrackLabel[] = "system_audio"; |
| |
| WebrtcAudioStream::WebrtcAudioStream() {} |
| |
| WebrtcAudioStream::~WebrtcAudioStream() { |
| if (stream_) { |
| for (const auto& track : stream_->GetAudioTracks()) { |
| stream_->RemoveTrack(track.get()); |
| } |
| peer_connection_->RemoveStream(stream_.get()); |
| } |
| } |
| |
| void WebrtcAudioStream::Start( |
| scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner, |
| std::unique_ptr<AudioSource> audio_source, |
| WebrtcTransport* webrtc_transport) { |
| DCHECK(webrtc_transport); |
| |
| source_adapter_ = |
| new rtc::RefCountedObject<WebrtcAudioSourceAdapter>(audio_task_runner); |
| source_adapter_->Start(std::move(audio_source)); |
| |
| scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_connection_factory( |
| webrtc_transport->peer_connection_factory()); |
| peer_connection_ = webrtc_transport->peer_connection(); |
| |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track = |
| peer_connection_factory->CreateAudioTrack(kAudioTrackLabel, |
| source_adapter_.get()); |
| |
| stream_ = peer_connection_factory->CreateLocalMediaStream(kAudioStreamLabel); |
| |
| // AddTrack() may fail only if there is another track with the same name, |
| // which is impossible because it's a brand new stream. |
| bool result = stream_->AddTrack(audio_track.get()); |
| DCHECK(result); |
| |
| // AddStream() may fail if there is another stream with the same name or when |
| // the PeerConnection is closed, neither is expected. |
| result = peer_connection_->AddStream(stream_.get()); |
| DCHECK(result); |
| } |
| |
| void WebrtcAudioStream::Pause(bool pause) { |
| source_adapter_->Pause(pause); |
| } |
| |
| } // namespace protocol |
| } // namespace remoting |