| Index: webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
|
| diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
|
| deleted file mode 100644
|
| index 675af70b4575fbc17508375f37b8ba5e0c79db4c..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
|
| +++ /dev/null
|
| @@ -1,63 +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/vad/vad_audio_proc.h"
|
| -
|
| -#include <math.h>
|
| -#include <stdio.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) {
|
| - 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};
|
| - 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
|
|
|