Make AudioProcessing::GetConfig() pure virtual

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 542 matching lines @@
   virtual EchoCancellation* echo_cancellation() const = 0;
   virtual EchoControlMobile* echo_control_mobile() const = 0;
   virtual GainControl* gain_control() const = 0;
   // TODO(peah): Deprecate this API call.
   virtual HighPassFilter* high_pass_filter() const = 0;
   virtual LevelEstimator* level_estimator() const = 0;
   virtual NoiseSuppression* noise_suppression() const = 0;
   virtual VoiceDetection* voice_detection() const = 0;
 
   // Returns the last applied configuration.
-  // TODO(henrik.lundin) Make this method pure virtual when downstream
-  // dependencies have been fixed.
-  virtual AudioProcessing::Config GetConfig() const;
+  virtual AudioProcessing::Config GetConfig() const = 0;
 
   enum Error {
     // Fatal errors.
     kNoError = 0,
     kUnspecifiedError = -1,
     kCreationFailedError = -2,
     kUnsupportedComponentError = -3,
     kUnsupportedFunctionError = -4,
     kNullPointerError = -5,
     kBadParameterError = -6,
@@ skipping 500 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_