Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

webrtc/media/engine/fakewebrtccall.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.
  */
 
@@ skipping 29 matching lines @@
     int payload_type = -1;
     int payload_frequency = -1;
     int event_code = 0;
     int duration_ms = 0;
   };
 
   explicit FakeAudioSendStream(
       int id, const webrtc::AudioSendStream::Config& config);
 
   int id() const { return id_; }
-  const webrtc::AudioSendStream::Config& GetConfig() const;
+  const webrtc::AudioSendStream::Config& GetConfig() const override;
+  webrtc::RtpState GetRtpState() const override;
+  const webrtc::TimeInterval& GetActiveLifetime() const override;
   void SetStats(const webrtc::AudioSendStream::Stats& stats);
   TelephoneEvent GetLatestTelephoneEvent() const;
   bool IsSending() const { return sending_; }
   bool muted() const { return muted_; }
 
  private:
   // webrtc::AudioSendStream implementation.
   void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
 
   void Start() override { sending_ = true; }
@@ skipping 259 matching lines @@
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 
   int audio_transport_overhead_;
   int video_transport_overhead_;
 };
 
 }  // namespace cricket
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_