Pull the Voice Activity Detector out from the AGC

webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc

Index: webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc
diff --git a/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc b/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc
deleted file mode 100644
index 9534aec2ec3ea0284bdd9786bc920f22a773337f..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- *  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 <math.h>
-#include <stdio.h>
-
-#include "gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) {
-  AgcAudioProc 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 };
-  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