blob: a7364626fd10da43fead058db19afc6526c691dc [file] [log] [blame]
/*
* Copyright (c) 2013 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 "modules/audio_processing/transient/transient_detector.h"
#include <memory>
#include <string>
#include "modules/audio_processing/transient/common.h"
#include "modules/audio_processing/transient/file_utils.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/file_wrapper.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
static const int kSampleRatesHz[] = {ts::kSampleRate8kHz, ts::kSampleRate16kHz,
ts::kSampleRate32kHz,
ts::kSampleRate48kHz};
static const size_t kNumberOfSampleRates =
sizeof(kSampleRatesHz) / sizeof(*kSampleRatesHz);
// This test is for the correctness of the transient detector.
// Checks the results comparing them with the ones stored in the detect files in
// the directory: resources/audio_processing/transient/
// The files contain all the results in double precision (Little endian).
// The audio files used with different sample rates are stored in the same
// directory.
#if defined(WEBRTC_IOS)
TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) {
#else
TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
#endif
for (size_t i = 0; i < kNumberOfSampleRates; ++i) {
int sample_rate_hz = kSampleRatesHz[i];
// Prepare detect file.
rtc::StringBuilder detect_file_name;
detect_file_name << "audio_processing/transient/detect"
<< (sample_rate_hz / 1000) << "kHz";
FileWrapper detect_file = FileWrapper::OpenReadOnly(
test::ResourcePath(detect_file_name.str(), "dat"));
bool file_opened = detect_file.is_open();
ASSERT_TRUE(file_opened) << "File could not be opened.\n"
<< detect_file_name.str().c_str();
// Prepare audio file.
rtc::StringBuilder audio_file_name;
audio_file_name << "audio_processing/transient/audio"
<< (sample_rate_hz / 1000) << "kHz";
FileWrapper audio_file = FileWrapper::OpenReadOnly(
test::ResourcePath(audio_file_name.str(), "pcm"));
// Create detector.
TransientDetector detector(sample_rate_hz);
const size_t buffer_length = sample_rate_hz * ts::kChunkSizeMs / 1000;
std::unique_ptr<float[]> buffer(new float[buffer_length]);
const float kTolerance = 0.02f;
size_t frames_read = 0;
while (ReadInt16FromFileToFloatBuffer(&audio_file, buffer_length,
buffer.get()) == buffer_length) {
++frames_read;
float detector_value =
detector.Detect(buffer.get(), buffer_length, NULL, 0);
double file_value;
ASSERT_EQ(1u, ReadDoubleBufferFromFile(&detect_file, 1, &file_value))
<< "Detect test file is malformed.\n";
// Compare results with data from the matlab test file.
EXPECT_NEAR(file_value, detector_value, kTolerance)
<< "Frame: " << frames_read;
}
detect_file.Close();
audio_file.Close();
}
}
} // namespace webrtc