blob: 0afed84c35dc2bf1dc7a7527e4265ac38caa9dfb [file] [log] [blame]
* 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 "modules/audio_processing/vad/vad_audio_proc.h"
#include <math.h>
#include <stdio.h>
#include <string>
#include "modules/audio_processing/vad/common.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.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 (size_t n = 0; n < features.num_frames; n++)
EXPECT_NEAR(features.spectral_peak[n], sp[n], 3);
} // namespace webrtc