blob: cd76a9557a650654e52d84da941e3e1bacbe747a [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 "gflags/gflags.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/test/gtest.h"
using std::complex;
namespace webrtc {
namespace {
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 file.");
// void function for gtest
void void_main(int argc, char* argv[]) {
google::SetUsageMessage(
"\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
google::ParseCommandLineFlags(&argc, &argv, true);
WavReader in_file(FLAGS_clear_file);
WavReader noise_file(FLAGS_noise_file);
WavWriter out_file(FLAGS_out_file, in_file.sample_rate(),
in_file.num_channels());
rtc::CriticalSection crit;
NoiseSuppressionImpl ns(&crit);
IntelligibilityEnhancer enh(in_file.sample_rate(), in_file.num_channels(), 1u,
NoiseSuppressionImpl::num_noise_bins());
ns.Initialize(noise_file.num_channels(), noise_file.sample_rate());
ns.Enable(true);
const size_t in_samples = noise_file.sample_rate() / 100;
const size_t noise_samples = noise_file.sample_rate() / 100;
std::vector<float> in(in_samples * in_file.num_channels());
std::vector<float> noise(noise_samples * noise_file.num_channels());
ChannelBuffer<float> in_buf(in_samples, in_file.num_channels());
ChannelBuffer<float> noise_buf(noise_samples, noise_file.num_channels());
AudioBuffer capture_audio(noise_samples, noise_file.num_channels(),
noise_samples, noise_file.num_channels(),
noise_samples);
AudioBuffer render_audio(in_samples, in_file.num_channels(), in_samples,
in_file.num_channels(), in_samples);
StreamConfig noise_config(noise_file.sample_rate(),
noise_file.num_channels());
StreamConfig in_config(in_file.sample_rate(), in_file.num_channels());
while (in_file.ReadSamples(in.size(), in.data()) == in.size() &&
noise_file.ReadSamples(noise.size(), noise.data()) == noise.size()) {
FloatS16ToFloat(noise.data(), noise.size(), noise.data());
FloatS16ToFloat(in.data(), in.size(), in.data());
Deinterleave(in.data(), in_buf.num_frames(), in_buf.num_channels(),
in_buf.channels());
Deinterleave(noise.data(), noise_buf.num_frames(), noise_buf.num_channels(),
noise_buf.channels());
capture_audio.CopyFrom(noise_buf.channels(), noise_config);
render_audio.CopyFrom(in_buf.channels(), in_config);
ns.AnalyzeCaptureAudio(&capture_audio);
ns.ProcessCaptureAudio(&capture_audio);
enh.SetCaptureNoiseEstimate(ns.NoiseEstimate(), 1);
enh.ProcessRenderAudio(&render_audio);
render_audio.CopyTo(in_config, in_buf.channels());
Interleave(in_buf.channels(), in_buf.num_frames(), in_buf.num_channels(),
in.data());
FloatToFloatS16(in.data(), in.size(), in.data());
out_file.WriteSamples(in.data(), in.size());
}
}
} // namespace
} // namespace webrtc
int main(int argc, char* argv[]) {
webrtc::void_main(argc, argv);
return 0;
}