Cleaned up and revised the handling of resampling and bandsplitting in APM and

webrtc/modules/audio_processing/gain_control_impl.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 25 matching lines @@
   int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
 
   void Initialize(size_t num_proc_channels, int sample_rate_hz);
 
   // GainControl implementation.
   bool is_enabled() const override;
   int stream_analog_level() override;
   bool is_limiter_enabled() const override;
   Mode mode() const override;
 
-  // Checks whether the module is enabled. Must only be
-  // called from the render side of APM as otherwise
-  // deadlocks may occur.
-  bool is_enabled_render_side_query() const;
-
   // Reads render side data that has been queued on the render call.
   void ReadQueuedRenderData();
 
   int compression_gain_db() const override;
 
  private:
   class GainController;
 
   // GainControl implementation.
   int Enable(bool enable) override;
@@ skipping 39 matching lines @@
   std::vector<std::unique_ptr<GainController>> gain_controllers_;
 
   rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
   rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_