Allow an external audio processing module to be used in WebRTC

webrtc/audio/audio_state.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include "webrtc/audio/audio_state.h"
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 
 namespace webrtc {
 namespace internal {
 
 AudioState::AudioState(const AudioState::Config& config)
     : config_(config),
       voe_base_(config.voice_engine),
       audio_transport_proxy_(voe_base_->audio_transport(),
-                             voe_base_->audio_processing(),
+                             config_.audio_processing.get(),
                              config_.audio_mixer) {
   process_thread_checker_.DetachFromThread();
   RTC_DCHECK(config_.audio_mixer);
 
   // Only one AudioState should be created per VoiceEngine.
   RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
 
   auto* const device = voe_base_->audio_device_module();
   RTC_DCHECK(device);
 
@@ skipping 52 matching lines @@
     typing_noise_detected_ = false;
   }
 }
 }  // namespace internal
 
 rtc::scoped_refptr<AudioState> AudioState::Create(
     const AudioState::Config& config) {
   return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
 }
 }  // namespace webrtc