Allow an external audio processing module to be used in WebRTC

webrtc/pc/peerconnectioninterface_unittest.cc

 /*
  *  Copyright 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 12 matching lines @@
 #include "webrtc/api/test/fakeconstraints.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/sctp/sctptransportinternal.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/mediastream.h"
 #include "webrtc/pc/peerconnection.h"
 #include "webrtc/pc/streamcollection.h"
 #include "webrtc/pc/test/fakertccertificategenerator.h"
 #include "webrtc/pc/test/fakevideotracksource.h"
 #include "webrtc/pc/test/mockpeerconnectionobservers.h"
 #include "webrtc/pc/test/testsdpstrings.h"
@@ skipping 600 matching lines @@
 class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
  public:
   static rtc::scoped_refptr<PeerConnectionFactoryForTest>
   CreatePeerConnectionFactoryForTest() {
     auto audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
     auto audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
 
     auto media_engine = std::unique_ptr<cricket::MediaEngineInterface>(
         cricket::WebRtcMediaEngineFactory::Create(
             nullptr, audio_encoder_factory, audio_decoder_factory, nullptr,
-            nullptr, nullptr));
+            nullptr, nullptr, webrtc::AudioProcessing::Create()));
 
     std::unique_ptr<webrtc::CallFactoryInterface> call_factory =
         webrtc::CreateCallFactory();
 
     std::unique_ptr<webrtc::RtcEventLogFactoryInterface> event_log_factory =
         webrtc::CreateRtcEventLogFactory();
 
     return new rtc::RefCountedObject<PeerConnectionFactoryForTest>(
         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
         nullptr, audio_encoder_factory, audio_decoder_factory, nullptr, nullptr,
@@ skipping 3030 matching lines @@
   EXPECT_NE(a, f);
 
   PeerConnectionInterface::RTCConfiguration g;
   g.disable_ipv6 = true;
   EXPECT_NE(a, g);
 
   PeerConnectionInterface::RTCConfiguration h(
       PeerConnectionInterface::RTCConfigurationType::kAggressive);
   EXPECT_NE(a, h);
 }