Allow an external audio processing module to be used in WebRTC

webrtc/voice_engine/voe_codec_unittest.cc

 /*
  *  Copyright (c) 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.
  */
 
 #include <memory>
 
 #include "webrtc/voice_engine/include/voe_codec.h"
 
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 namespace voe {
 namespace {
 
 TEST(VoECodecInst, TestCompareCodecInstances) {
   CodecInst codec1, codec2;
@@ skipping 46 matching lines @@
 }
 
 // This is a regression test for
 // https://bugs.chromium.org/p/webrtc/issues/detail?id=6020
 // The Opus DTX setting was being forgotten after unrelated VoE calls.
 TEST(VoECodecInst, RememberOpusDtxAfterSettingChange) {
   VoiceEngine* voe(VoiceEngine::Create());
   VoEBase* base(VoEBase::GetInterface(voe));
   VoECodec* voe_codec(VoECodec::GetInterface(voe));
   std::unique_ptr<FakeAudioDeviceModule> adm(new FakeAudioDeviceModule);
+  std::unique_ptr<AudioProcessing> apm(AudioProcessing::Create());
 
-  base->Init(adm.get());
+  base->Init(adm.get(), apm.get());
 
   CodecInst codec = {111, "opus", 48000, 960, 1, 32000};
 
   int channel = base->CreateChannel();
 
   bool DTX = false;
 
   EXPECT_EQ(0, voe_codec->SetSendCodec(channel, codec));
   EXPECT_EQ(0, voe_codec->SetOpusDtx(channel, true));
   EXPECT_EQ(0, voe_codec->SetFECStatus(channel, true));
   EXPECT_EQ(0, voe_codec->GetOpusDtxStatus(channel, &DTX));
   EXPECT_TRUE(DTX);
 
   base->DeleteChannel(channel);
   base->Terminate();
   base->Release();
   voe_codec->Release();
   VoiceEngine::Delete(voe);
 }
 
 }  // namespace
 }  // namespace voe
 }  // namespace webrtc