Allow an external audio processing module to be used in WebRTC

webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

Index: webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index 6fbf5f1297f8cc3254cd8c34625b0d00aac869ef..9c3fc2ec41359769da4ee9209b20aec15e577f72 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -55,6 +55,9 @@ ConferenceTransport::ConferenceTransport()
   local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
   local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);
 
+  local_apm_ = webrtc::AudioProcessing::Create();
+  local_base_->Init(nullptr, local_apm_.get(), nullptr);
+
   // In principle, we can use one VoiceEngine to achieve the same goal. Well, in
   // here, we use two engines to make it more like reality.
   remote_voe_ = webrtc::VoiceEngine::Create();
@@ -64,7 +67,9 @@ ConferenceTransport::ConferenceTransport()
   remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
   remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);
 
-  EXPECT_EQ(0, local_base_->Init());
+  remote_apm_.reset(webrtc::AudioProcessing::Create());
+  remote_base_->Init(nullptr, remote_apm_.get(), nullptr);
+
   local_sender_ = local_base_->CreateChannel();
   static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
       ->GetChannelProxy(local_sender_)
@@ -74,10 +79,8 @@ ConferenceTransport::ConferenceTransport()
   EXPECT_EQ(0, local_rtp_rtcp_->
       SetSendAudioLevelIndicationStatus(local_sender_, true,
                                         kAudioLevelHeaderId));
-
   EXPECT_EQ(0, local_base_->StartSend(local_sender_));
 
-  EXPECT_EQ(0, remote_base_->Init());
   reflector_ = remote_base_->CreateChannel();
   static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
       ->GetChannelProxy(reflector_)