| // 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. |
| |
| #ifndef REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_ |
| #define REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_ |
| |
| #include <memory> |
| |
| #include "base/memory/ref_counted.h" |
| #include "remoting/protocol/audio_stream.h" |
| #include "third_party/webrtc/api/scoped_refptr.h" |
| |
| namespace base { |
| class SingleThreadTaskRunner; |
| } // namespace webrtc |
| |
| namespace webrtc { |
| class PeerConnectionInterface; |
| } // namespace webrtc |
| |
| namespace remoting { |
| namespace protocol { |
| |
| class AudioSource; |
| class WebrtcAudioSourceAdapter; |
| class WebrtcTransport; |
| |
| class WebrtcAudioStream : public AudioStream { |
| public: |
| WebrtcAudioStream(); |
| |
| WebrtcAudioStream(const WebrtcAudioStream&) = delete; |
| WebrtcAudioStream& operator=(const WebrtcAudioStream&) = delete; |
| |
| ~WebrtcAudioStream() override; |
| |
| void Start(scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner, |
| std::unique_ptr<AudioSource> audio_source, |
| WebrtcTransport* webrtc_transport); |
| |
| // AudioStream interface. |
| void Pause(bool pause) override; |
| |
| private: |
| scoped_refptr<WebrtcAudioSourceAdapter> source_adapter_; |
| |
| scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| }; |
| |
| } // namespace protocol |
| } // namespace remoting |
| |
| #endif // REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_ |