Add AudioSendStream to Call.

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
new file mode 100644
index 0000000000000000000000000000000000000000..54046fcea96d3d3b8d145c9b480f7224afe11b37
--- /dev/null
+++ b/webrtc/audio/audio_send_stream.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
+#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
+
+#include "webrtc/audio_send_stream.h"
+
+namespace webrtc {
+namespace internal {
+
+class AudioSendStream : public webrtc::AudioSendStream {
+ public:
+  explicit AudioSendStream(const webrtc::AudioSendStream::Config& config);
+  ~AudioSendStream() override;
+
+  // webrtc::SendStream implementation.
+  void Start() override;
+  void Stop() override;
+  void SignalNetworkState(NetworkState state) override;
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
+
+  // webrtc::AudioSendStream implementation.
+  webrtc::AudioSendStream::Stats GetStats() const override;
+
+  const webrtc::AudioSendStream::Config& config() const {
+    return config_;
+  }
+
+ private:
+  const webrtc::AudioSendStream::Config config_;
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_