Fix chromium-style warnings in webrtc/sound/.

webrtc/sound/soundinputstreaminterface.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_SOUNDINPUTSTREAMINTERFACE_H_
 #define WEBRTC_SOUND_SOUNDINPUTSTREAMINTERFACE_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 
 namespace rtc {
 
 // Interface for consuming an input stream from a recording device.
 // Semantics and thread-safety of StartReading()/StopReading() are the same as
 // for rtc::Worker.
 class SoundInputStreamInterface {
  public:
-  virtual ~SoundInputStreamInterface() {}
+  virtual ~SoundInputStreamInterface();
 
   // Starts the reading of samples on the current thread.
   virtual bool StartReading() = 0;
   // Stops the reading of samples.
   virtual bool StopReading() = 0;
 
   // Retrieves the current input volume for this stream. Nominal range is
   // defined by SoundSystemInterface::k(Max|Min)Volume, but values exceeding the
   // max may be possible in some implementations. This call retrieves the actual
   // volume currently in use by the OS, not a cached value from a previous
@@ skipping 15 matching lines @@
   // Notifies the consumer of new data read from the device.
   // The first parameter is a pointer to the data read, and is only valid for
   // the duration of the call.
   // The second parameter is the amount of data read in bytes (i.e., the valid
   // length of the memory pointed to).
   // The 3rd parameter is the stream that is issuing the callback.
   sigslot::signal3<const void *, size_t,
       SoundInputStreamInterface *> SignalSamplesRead;
 
  protected:
-  SoundInputStreamInterface() {}
+  SoundInputStreamInterface();
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(SoundInputStreamInterface);
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_