Clean up the Config to enable 48kHz support in AudioProcessing

webrtc/modules/audio_processing/audio_processing_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 176 matching lines @@
       use_new_agc_(config.Get<ExperimentalAgc>().enabled),
 #endif
       agc_startup_min_volume_(config.Get<ExperimentalAgc>().startup_min_volume),
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
       transient_suppressor_enabled_(false),
 #else
       transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
 #endif
       beamformer_enabled_(config.Get<Beamforming>().enabled),
       beamformer_(beamformer),
-      array_geometry_(config.Get<Beamforming>().array_geometry),
-      supports_48kHz_(config.Get<AudioProcessing48kHzSupport>().enabled) {
+      array_geometry_(config.Get<Beamforming>().array_geometry) {
   echo_cancellation_ = new EchoCancellationImpl(this, crit_);
   component_list_.push_back(echo_cancellation_);
 
   echo_control_mobile_ = new EchoControlMobileImpl(this, crit_);
   component_list_.push_back(echo_control_mobile_);
 
   gain_control_ = new GainControlImpl(this, crit_);
   component_list_.push_back(gain_control_);
 
   high_pass_filter_ = new HighPassFilterImpl(this, crit_);
@@ skipping 134 matching lines @@
     return kBadNumberChannelsError;
   }
 
   fwd_in_format_.set(input_sample_rate_hz, num_input_channels);
   fwd_out_format_.set(output_sample_rate_hz, num_output_channels);
   rev_in_format_.set(reverse_sample_rate_hz, num_reverse_channels);
 
   // We process at the closest native rate >= min(input rate, output rate)...
   int min_proc_rate = std::min(fwd_in_format_.rate(), fwd_out_format_.rate());
   int fwd_proc_rate;
-  if (supports_48kHz_ && min_proc_rate > kSampleRate32kHz) {
+  if (min_proc_rate > kSampleRate32kHz) {
     fwd_proc_rate = kSampleRate48kHz;
   } else if (min_proc_rate > kSampleRate16kHz) {
     fwd_proc_rate = kSampleRate32kHz;
   } else if (min_proc_rate > kSampleRate8kHz) {
     fwd_proc_rate = kSampleRate16kHz;
   } else {
     fwd_proc_rate = kSampleRate8kHz;
   }
   // ...with one exception.
   if (echo_control_mobile_->is_enabled() && min_proc_rate > kSampleRate16kHz) {
@@ skipping 742 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc