Removed inline definitions and added destructors to fix chromium-style.

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 127 matching lines @@
 struct ExperimentalNs {
   ExperimentalNs() : enabled(false) {}
   explicit ExperimentalNs(bool enabled) : enabled(enabled) {}
   static const ConfigOptionID identifier = ConfigOptionID::kExperimentalNs;
   bool enabled;
 };
 
 // Use to enable beamforming. Must be provided through the constructor. It will
 // have no impact if used with AudioProcessing::SetExtraOptions().
 struct Beamforming {
-  Beamforming()
-      : enabled(false),
-        array_geometry(),
-        target_direction(
-            SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f)) {}
-  Beamforming(bool enabled, const std::vector<Point>& array_geometry)
-      : Beamforming(enabled,
-                    array_geometry,
-                    SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f)) {
-  }
+  Beamforming();
+  Beamforming(bool enabled, const std::vector<Point>& array_geometry);
   Beamforming(bool enabled,
               const std::vector<Point>& array_geometry,
-              SphericalPointf target_direction)
-      : enabled(enabled),
-        array_geometry(array_geometry),
-        target_direction(target_direction) {}
+              SphericalPointf target_direction);
+  ~Beamforming();
+
   static const ConfigOptionID identifier = ConfigOptionID::kBeamforming;
   const bool enabled;
   const std::vector<Point> array_geometry;
   const SphericalPointf target_direction;
 };
 
 // Use to enable intelligibility enhancer in audio processing.
 //
 // Note: If enabled and the reverse stream has more than one output channel,
 // the reverse stream will become an upmixed mono signal.
@@ skipping 269 matching lines @@
   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);
   }
 
   // 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) {
-      return -1;
-  }
+  virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle);
 
   // 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 519 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_