Enable cpplint and fix cpplint errors in webrtc/*audio

webrtc/common_audio/audio_converter.h

 /*
  *  Copyright (c) 2014 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 13 matching lines @@
 // The source and destination chunks have the same duration in time; specifying
 // the number of frames is equivalent to specifying the sample rates.
 class AudioConverter {
  public:
   // Returns a new AudioConverter, which will use the supplied format for its
   // lifetime. Caller is responsible for the memory.
   static std::unique_ptr<AudioConverter> Create(size_t src_channels,
                                                 size_t src_frames,
                                                 size_t dst_channels,
                                                 size_t dst_frames);
-  virtual ~AudioConverter() {};
+  virtual ~AudioConverter() {}
 
   // Convert |src|, containing |src_size| samples, to |dst|, having a sample
   // capacity of |dst_capacity|. Both point to a series of buffers containing
   // the samples for each channel. The sizes must correspond to the format
   // passed to Create().
   virtual void Convert(const float* const* src, size_t src_size,
                        float* const* dst, size_t dst_capacity) = 0;
 
   size_t src_channels() const { return src_channels_; }
   size_t src_frames() const { return src_frames_; }
@@ skipping 13 matching lines @@
   const size_t src_frames_;
   const size_t dst_channels_;
   const size_t dst_frames_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_