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/*
* 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 <string>
#include <iostream>
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
int main(int argc, char* argv[]) {
if (argc != 5) {
std::cout << "Usage: " << argv[0] <<
" channels output_type <input file name> <output file name> "
<< std::endl << std::endl;
std::cout << "Where channels can be 1 (mono) or 2 (interleaved stereo),";
std::cout << " outputs can be 1 (classification (boolean)) or 2";
std::cout << " (classification and music probability (float)),"
<< std::endl;
std::cout << "and the sampling frequency is assumed to be 48 kHz."
<< std::endl;
return -1;
}
const int kFrameSizeSamples = 960;
int channels = atoi(argv[1]);
if (channels < 1 || channels > 2) {
std::cout << "Disallowed number of channels " << channels << std::endl;
return -1;
}
int outputs = atoi(argv[2]);
if (outputs < 1 || outputs > 2) {
std::cout << "Disallowed number of outputs " << outputs << std::endl;
return -1;
}
const int data_size = channels * kFrameSizeSamples;
webrtc::scoped_ptr<int16_t[]> in(new int16_t[data_size]);
std::string input_filename = argv[3];
std::string output_filename = argv[4];
std::cout << "Input file: " << input_filename << std::endl;
std::cout << "Output file: " << output_filename << std::endl;
FILE* in_file = fopen(input_filename.c_str(), "rb");
if (!in_file) {
std::cout << "Cannot open input file " << input_filename << std::endl;
return -1;
}
FILE* out_file = fopen(output_filename.c_str(), "wb");
if (!out_file) {
std::cout << "Cannot open output file " << output_filename << std::endl;
return -1;
}
webrtc::AudioClassifier classifier;
int frame_counter = 0;
int music_counter = 0;
while (fread(in.get(), sizeof(*in.get()),
data_size, in_file) == (size_t) data_size) {
bool is_music = classifier.Analysis(in.get(), data_size, channels);
if (!fwrite(&is_music, sizeof(is_music), 1, out_file)) {
std::cout << "Error writing." << std::endl;
return -1;
}
if (is_music) {
music_counter++;
}
std::cout << "frame " << frame_counter << " decision " << is_music;
if (outputs == 2) {
float music_prob = classifier.music_probability();
if (!fwrite(&music_prob, sizeof(music_prob), 1, out_file)) {
std::cout << "Error writing." << std::endl;
return -1;
}
std::cout << " music prob " << music_prob;
}
std::cout << std::endl;
frame_counter++;
}
std::cout << frame_counter << " frames processed." << std::endl;
if (frame_counter > 0) {
float music_percentage = music_counter / static_cast<float>(frame_counter);
std::cout << music_percentage << " percent music." << std::endl;
}
fclose(in_file);
fclose(out_file);
return 0;
}
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