Allow an external audio processing module to be used in WebRTC

webrtc/audio/audio_state.h

 /*
  *  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.
  */
 
@@ skipping 10 matching lines @@
 
 namespace webrtc {
 namespace internal {
 
 class AudioState final : public webrtc::AudioState,
                          public webrtc::VoiceEngineObserver {
  public:
   explicit AudioState(const AudioState::Config& config);
   ~AudioState() override;
 
+  // TODO(peah): Remove the conditional when upstream dependencies have properly
+  // been resolved.
+  AudioProcessing* audio_processing() override {
+    return config_.audio_processing ? config_.audio_processing.get()
+                                    : voe_base_->audio_processing();
+  }
+
   VoiceEngine* voice_engine();
-
   rtc::scoped_refptr<AudioMixer> mixer();
   bool typing_noise_detected() const;
 
  private:
   // rtc::RefCountInterface implementation.
   int AddRef() const override;
   int Release() const override;
 
   // webrtc::VoiceEngineObserver implementation.
   void CallbackOnError(int channel_id, int err_code) override;
@@ skipping 16 matching lines @@
   // Transports mixed audio from the mixer to the audio device and
   // recorded audio to the VoE AudioTransport.
   AudioTransportProxy audio_transport_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_STATE_H_