Allow an external audio processing module to be used in WebRTC

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 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.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
@@ skipping 128 matching lines @@
     VoiceEngine* voice_engine;
     VoEBase* base;
     int channel_id;
   };
 
   void CreateVoiceEngines();
   void DestroyVoiceEngines();
 
   VoiceEngineState voe_send_;
   VoiceEngineState voe_recv_;
+  rtc::scoped_refptr<AudioProcessing> apm_send_;
+  rtc::scoped_refptr<AudioProcessing> apm_recv_;
 
   // The audio devices must outlive the voice engines.
   std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
   std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
 };
 
 class BaseTest : public RtpRtcpObserver {
  public:
   BaseTest();
   explicit BaseTest(unsigned int timeout_ms);
@@ skipping 61 matching lines @@
   EndToEndTest();
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_