Fix chromium-style warnings in webrtc/sound/.

webrtc/sound/soundoutputstreaminterface.h

 /*
  *  Copyright 2004 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_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
 #define WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 
 namespace rtc {
 
 // Interface for outputting a stream to a playback device.
 // Semantics and thread-safety of EnableBufferMonitoring()/
 // DisableBufferMonitoring() are the same as for rtc::Worker.
 class SoundOutputStreamInterface {
  public:
-  virtual ~SoundOutputStreamInterface() {}
+  virtual ~SoundOutputStreamInterface();
 
   // Enables monitoring the available buffer space on the current thread.
   virtual bool EnableBufferMonitoring() = 0;
   // Disables the monitoring.
   virtual bool DisableBufferMonitoring() = 0;
 
   // Write the given samples to the devices. If currently monitoring then this
   // may only be called from the monitoring thread.
   virtual bool WriteSamples(const void *sample_data,
                             size_t size) = 0;
@@ skipping 19 matching lines @@
 
   // Notifies the producer of the available buffer space for writes.
   // It fires continuously as long as the space is greater than zero.
   // The first parameter is the amount of buffer space available for data to
   // be written (i.e., the maximum amount of data that can be written right now
   // with WriteSamples() without blocking).
   // The 2nd parameter is the stream that is issuing the callback.
   sigslot::signal2<size_t, SoundOutputStreamInterface *> SignalBufferSpace;
 
  protected:
-  SoundOutputStreamInterface() {}
+  SoundOutputStreamInterface();
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(SoundOutputStreamInterface);
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_