Allow an external audio processing module to be used in WebRTC

webrtc/call/audio_state.h

Index: webrtc/call/audio_state.h
diff --git a/webrtc/call/audio_state.h b/webrtc/call/audio_state.h
index 2c26a1749b5b62df61cef8aba5d606d23d1b92a9..32b7ada3f80505625f300e62569726d6f5a641cb 100644
--- a/webrtc/call/audio_state.h
+++ b/webrtc/call/audio_state.h
@@ -16,6 +16,7 @@
 
 namespace webrtc {
 
+class AudioProcessing;
 class VoiceEngine;
 
 // WORK IN PROGRESS
@@ -36,8 +37,13 @@ class AudioState : public rtc::RefCountInterface {
     // The audio mixer connected to active receive streams. One per
     // AudioState.
     rtc::scoped_refptr<AudioMixer> audio_mixer;
+
+    // The audio processing module.
+    rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
   };
 
+  virtual AudioProcessing* audio_processing() = 0;
+
   // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
   static rtc::scoped_refptr<AudioState> Create(
       const AudioState::Config& config);