Made the AudioProcessing class a pure interface.

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 2012 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 436 matching lines @@
   // <= 0, no limit will be used.
   static const size_t kMaxFilenameSize = 1024;
   virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
                                   int64_t max_log_size_bytes) = 0;
 
   // Same as above but uses an existing file handle. Takes ownership
   // of |handle| and closes it at StopDebugRecording().
   virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
 
   // TODO(ivoc): Remove this function after Chrome stops using it.
-  int StartDebugRecording(FILE* handle) {
-    return StartDebugRecording(handle, -1);
-  }
+  virtual int StartDebugRecording(FILE* handle) = 0;
 
   // Same as above but uses an existing PlatformFile handle. Takes ownership
   // of |handle| and closes it at StopDebugRecording().
   // TODO(xians): Make this interface pure virtual.
-  virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle);
+  virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) = 0;
 
   // Stops recording debugging information, and closes the file. Recording
   // cannot be resumed in the same file (without overwriting it).
   virtual int StopDebugRecording() = 0;
 
   // Use to send UMA histograms at end of a call. Note that all histogram
   // specific member variables are reset.
   virtual void UpdateHistogramsOnCallEnd() = 0;
 
   // These provide access to the component interfaces and should never return
@@ skipping 525 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_