Only split into bands when the reverse stream is analyzed in the APM

webrtc/modules/audio_processing/audio_processing_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 189 matching lines @@
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeEchoControlMobile()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   int InitializeLocked(const ProcessingConfig& config)
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
   // Capture-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   bool output_copy_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
-  bool is_data_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool is_fwd_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   bool fwd_synthesis_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   bool fwd_analysis_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   // Render-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   // TODO(ekm): Remove once all clients updated to new interface.
   int AnalyzeReverseStreamLocked(const float* const* src,
                                  const StreamConfig& input_config,
                                  const StreamConfig& output_config)
@@ skipping 120 matching lines @@
 
   struct ApmRenderState {
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_