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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.
*/
//
// Command line tool for speech intelligibility enhancement. Provides for
// running and testing intelligibility_enhancer as an independent process.
// Use --help for options.
//
#include <stdint.h>
#include <stdlib.h>
#include <sys/stat.h>
#include <sys/types.h>
#include <string>
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/real_fourier.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
using std::complex;
using webrtc::intelligibility::VarianceArray;
namespace webrtc {
namespace {
bool ValidateClearWindow(const char* flagname, int32_t value) {
return value > 0;
}
DEFINE_int32(clear_type,
webrtc::intelligibility::VarianceArray::kStepDecaying,
"Variance algorithm for clear data.");
DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data.");
DEFINE_int32(clear_window,
475,
"Window size for windowed variance for clear data.");
const bool clear_window_dummy =
google::RegisterFlagValidator(&FLAGS_clear_window, &ValidateClearWindow);
DEFINE_int32(sample_rate,
16000,
"Audio sample rate used in the input and output files.");
DEFINE_int32(ana_rate,
800,
"Analysis rate; gains recalculated every N blocks.");
DEFINE_int32(
var_rate,
2,
"Variance clear rate; history is forgotten every N gain recalculations.");
DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
DEFINE_string(out_file,
"proc_enhanced.wav",
"Enhanced output. Use '-' to "
"play through aplay immediately.");
const int kNumChannels = 1;
// void function for gtest
void void_main(int argc, char* argv[]) {
google::SetUsageMessage(
"\n\nVariance algorithm types are:\n"
" 0 - infinite/normal,\n"
" 1 - exponentially decaying,\n"
" 2 - rolling window.\n"
"\nInput files must be little-endian 16-bit signed raw PCM.\n");
google::ParseCommandLineFlags(&argc, &argv, true);
size_t samples; // Number of samples in input PCM file
size_t fragment_size; // Number of samples to process at a time
// to simulate APM stream processing
// Load settings and wav input.
fragment_size = FLAGS_sample_rate / 100; // Mirror real time APM chunk size.
// Duplicates chunk_length_ in
// IntelligibilityEnhancer.
struct stat in_stat, noise_stat;
ASSERT_EQ(stat(FLAGS_clear_file.c_str(), &in_stat), 0)
<< "Empty speech file.";
ASSERT_EQ(stat(FLAGS_noise_file.c_str(), &noise_stat), 0)
<< "Empty noise file.";
samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
WavReader in_file(FLAGS_clear_file);
std::vector<float> in_fpcm(samples);
in_file.ReadSamples(samples, &in_fpcm[0]);
WavReader noise_file(FLAGS_noise_file);
std::vector<float> noise_fpcm(samples);
noise_file.ReadSamples(samples, &noise_fpcm[0]);
// Run intelligibility enhancement.
IntelligibilityEnhancer::Config config;
config.sample_rate_hz = FLAGS_sample_rate;
config.var_type = static_cast<VarianceArray::StepType>(FLAGS_clear_type);
config.var_decay_rate = static_cast<float>(FLAGS_clear_alpha);
config.var_window_size = static_cast<size_t>(FLAGS_clear_window);
config.analysis_rate = FLAGS_ana_rate;
config.gain_change_limit = FLAGS_gain_limit;
IntelligibilityEnhancer enh(config);
// Slice the input into smaller chunks, as the APM would do, and feed them
// through the enhancer.
float* clear_cursor = &in_fpcm[0];
float* noise_cursor = &noise_fpcm[0];
for (size_t i = 0; i < samples; i += fragment_size) {
enh.AnalyzeCaptureAudio(&noise_cursor, FLAGS_sample_rate, kNumChannels);
enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels);
clear_cursor += fragment_size;
noise_cursor += fragment_size;
}
if (FLAGS_out_file.compare("-") == 0) {
const std::string temp_out_filename =
test::TempFilename(test::WorkingDir(), "temp_wav_file");
{
WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels);
out_file.WriteSamples(&in_fpcm[0], samples);
}
system(("aplay " + temp_out_filename).c_str());
system(("rm " + temp_out_filename).c_str());
} else {
WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
out_file.WriteSamples(&in_fpcm[0], samples);
}
}
} // namespace
} // namespace webrtc
int main(int argc, char* argv[]) {
webrtc::void_main(argc, argv);
return 0;
}
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