Index: remoting/protocol/webrtc_audio_stream.h |
diff --git a/remoting/protocol/webrtc_audio_stream.h b/remoting/protocol/webrtc_audio_stream.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..6c5981a3aa44c045661037d50d24a4c45cadd1e2 |
--- /dev/null |
+++ b/remoting/protocol/webrtc_audio_stream.h |
@@ -0,0 +1,54 @@ |
+// 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/macros.h" |
+#include "base/memory/ref_counted.h" |
+#include "remoting/protocol/audio_stream.h" |
+ |
+namespace base { |
+class SingleThreadTaskRunner; |
+} // namespace webrtc |
+ |
+namespace webrtc { |
+class MediaStreamInterface; |
+class PeerConnectionInterface; |
+} // namespace webrtc |
+ |
+namespace remoting { |
+namespace protocol { |
+ |
+class AudioSource; |
+class WebrtcAudioSourceAdapter; |
+class WebrtcTransport; |
+ |
+class WebrtcAudioStream : public AudioStream { |
+ public: |
+ WebrtcAudioStream(); |
+ ~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_; |
+ scoped_refptr<webrtc::MediaStreamInterface> stream_; |
+ |
+ DISALLOW_COPY_AND_ASSIGN(WebrtcAudioStream); |
+}; |
+ |
+} // namespace protocol |
+} // namespace remoting |
+ |
+#endif // REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_ |