Made the AudioProcessing class a pure interface.

webrtc/modules/audio_processing/include/audio_processing.h

Index: webrtc/modules/audio_processing/include/audio_processing.h
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index b5d2aa22c0348cd4da2011fb7a67803d80862337..0fdbfd2d75cf48caa50bb8b2f8a86f5e156ee353 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -454,14 +454,12 @@ class AudioProcessing {
   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).