Allow an external audio processing module to be used in WebRTC

webrtc/voice_engine/test/auto_test/standard/codec_test.cc

 /*
  *  Copyright (c) 2011 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 <stdio.h>
 #include <string>
 
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 class CodecTest : public AfterStreamingFixture {
  protected:
   void SetUp() {
     memset(&codec_instance_, 0, sizeof(codec_instance_));
+    apm_ = webrtc::AudioProcessing::Create();
+    voe_base_->Init(nullptr, apm_.get(), nullptr);
   }
 
   void SetArbitrarySendCodec() {
     // Just grab the first codec.
     EXPECT_EQ(0, voe_codec_->GetCodec(0, codec_instance_));
     EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, codec_instance_));
   }
 
+  rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
   webrtc::CodecInst codec_instance_;
 };
 
 static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) {
   if (!STR_CASE_CMP(codec_instance->plname, "ilbc")) {
     if (packet_size == 160 || packet_size == 320) {
       codec_instance->rate = 15200;
     } else {
       codec_instance->rate = 13300;
     }
@@ skipping 164 matching lines @@
         TEST_LOG("%d ", packet_size);
         TEST_LOG_FLUSH;
         at_least_one_succeeded = true;
         Sleep(CODEC_TEST_TIME);
       }
     }
     TEST_LOG("\n");
     EXPECT_TRUE(at_least_one_succeeded);
   }
 }