Allow an external audio processing module to be used in WebRTC

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

 /*
  *  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 37 matching lines @@
       rtp_header_parser_(webrtc::RtpHeaderParser::Create()) {
   rtp_header_parser_->
       RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
                                  kAudioLevelHeaderId);
 
   local_voe_ = webrtc::VoiceEngine::Create();
   local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
   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();
   remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
   remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
   remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
   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_)
       ->RegisterLegacyReceiveCodecs();
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
   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_)
       ->RegisterLegacyReceiveCodecs();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
   EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
 
   thread_.Start();
   thread_.SetPriority(rtc::kHighPriority);
 }
@@ skipping 204 matching lines @@
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
   return true;
 }
 
 }  // namespace voetest
 }  // namespace webrtc