Define Stream base classes

webrtc/audio_receive_stream.h

 /*
  *  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_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/config.h"
+#include "webrtc/stream.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioDecoder;
 
-class AudioReceiveStream {
+class AudioReceiveStream : public ReceiveStream {
  public:
   struct Stats {};
 
   struct Config {
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
       // Synchronization source (stream identifier) to be received.
       uint32_t remote_ssrc = 0;
 
       // Sender SSRC used for sending RTCP (such as receiver reports).
       uint32_t local_ssrc = 0;
 
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
     // Decoders for every payload that we can receive. Call owns the
     // AudioDecoder instances once the Config is submitted to
     // Call::CreateReceiveStream().
     // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
     std::map<uint8_t, AudioDecoder*> decoder_map;
   };
 
   virtual Stats GetStats() const = 0;
-
- protected:
-  virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_