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

webrtc/modules/audio_processing/gain_control_impl.cc

 /*
  *  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 298 matching lines @@
 
     RTC_DCHECK(num_proc_channels_);
     RTC_DCHECK(sample_rate_hz_);
     Initialize(*num_proc_channels_, *sample_rate_hz_);
   } else {
     enabled_ = enable;
   }
   return AudioProcessing::kNoError;
 }
 
-bool GainControlImpl::is_enabled_render_side_query() const {
-  // TODO(peah): Add threadchecker.
-  rtc::CritScope cs(crit_render_);
-  return enabled_;
-}
-
 bool GainControlImpl::is_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return enabled_;
 }
 
 int GainControlImpl::set_mode(Mode mode) {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   if (MapSetting(mode) == -1) {
     return AudioProcessing::kBadParameterError;
@@ skipping 164 matching lines @@
   for (auto& gain_controller : gain_controllers_) {
     const int handle_error =
         WebRtcAgc_set_config(gain_controller->state(), config);
     if (handle_error != AudioProcessing::kNoError) {
       error = handle_error;
     }
   }
   return error;
 }
 }  // namespace webrtc