blob: bdc5a05920dd94c01b49524c4bcd0c46a50eea63 [file] [log] [blame]
/*
* Copyright (c) 2014 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 "webrtc/modules/audio_coding/neteq/audio_classifier.h"
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <memory>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
static const size_t kFrameSize = 960;
TEST(AudioClassifierTest, AllZeroInput) {
int16_t in_mono[kFrameSize] = {0};
// Test all-zero vectors and let the classifier converge from its default
// to the expected value.
AudioClassifier zero_classifier;
for (int i = 0; i < 100; ++i) {
zero_classifier.Analysis(in_mono, kFrameSize, 1);
}
EXPECT_TRUE(zero_classifier.is_music());
}
void RunAnalysisTest(const std::string& audio_filename,
const std::string& data_filename,
size_t channels) {
AudioClassifier classifier;
std::unique_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]);
bool is_music_ref;
FILE* audio_file = fopen(audio_filename.c_str(), "rb");
ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
<< std::endl;
FILE* data_file = fopen(data_filename.c_str(), "rb");
ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
<< std::endl;
while (fread(in.get(), sizeof(int16_t), channels * kFrameSize, audio_file) ==
channels * kFrameSize) {
bool is_music =
classifier.Analysis(in.get(), channels * kFrameSize, channels);
EXPECT_EQ(is_music, classifier.is_music());
ASSERT_EQ(1u, fread(&is_music_ref, sizeof(is_music_ref), 1, data_file));
EXPECT_EQ(is_music_ref, is_music);
}
fclose(audio_file);
fclose(data_file);
}
TEST(AudioClassifierTest, DoAnalysisMono) {
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
test::ResourcePath("short_mixed_mono_48_arm", "dat"),
1);
#else
RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
test::ResourcePath("short_mixed_mono_48", "dat"),
1);
#endif // WEBRTC_ARCH_ARM
}
TEST(AudioClassifierTest, DoAnalysisStereo) {
RunAnalysisTest(test::ResourcePath("short_mixed_stereo_48", "pcm"),
test::ResourcePath("short_mixed_stereo_48", "dat"),
2);
}
} // namespace webrtc