Pull the Voice Activity Detector out from the AGC

webrtc/modules/audio_processing/vad/vad_audio_proc_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.
  */
 
 // We don't test the value of pitch gain and lags as they are created by iSAC
 // routines. However, interpolation of pitch-gain and lags is in a separate
 // class and has its own unit-test.
 
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h"
 
 #include <math.h>
 #include <stdio.h>
 
-#include "gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) {
-  AgcAudioProc audioproc;
+  VadAudioProc audioproc;
 
   std::string peak_file_name =
       test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat");
   FILE* peak_file = fopen(peak_file_name.c_str(), "rb");
   ASSERT_TRUE(peak_file != NULL);
 
   std::string pcm_file_name =
       test::ResourcePath("audio_processing/agc/agc_audio", "pcm");
   FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb");
   ASSERT_TRUE(pcm_file != NULL);
 
   // Read 10 ms audio in each iteration.
   const size_t kDataLength = kLength10Ms;
-  int16_t data[kDataLength] = { 0 };
+  int16_t data[kDataLength] = {0};
   AudioFeatures features;
   double sp[kMaxNumFrames];
   while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) {
     audioproc.ExtractFeatures(data, kDataLength, &features);
     if (features.num_frames > 0) {
       ASSERT_LT(features.num_frames, kMaxNumFrames);
       // Read reference values.
       const size_t num_frames = features.num_frames;
       ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file));
       for (int n = 0; n < features.num_frames; n++)
         EXPECT_NEAR(features.spectral_peak[n], sp[n], 3);
     }
   }
 
   fclose(peak_file);
   fclose(pcm_file);
 }
 
 }  // namespace webrtc