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* 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 <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <memory>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc/agc.h"
#include "modules/audio_processing/transient/transient_suppressor_impl.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
ABSL_FLAG(std::string, in_file_name, "", "PCM file that contains the signal.");
"PCM file that contains the detection signal.");
"PCM file that contains the reference signal.");
"Time between each chunk of samples in milliseconds.");
"Sampling frequency of the signal in Hertz.");
"Sampling frequency of the detection signal in Hertz.");
ABSL_FLAG(int, num_channels, 1, "Number of channels.");
namespace webrtc {
const char kUsage[] =
"\nDetects and suppresses transients from file.\n\n"
"This application loads the signal from the in_file_name with a specific\n"
"num_channels and sample_rate_hz, the detection signal from the\n"
"detection_file_name with a specific detection_rate_hz, and the reference\n"
"signal from the reference_file_name with sample_rate_hz, divides them\n"
"into chunk_size_ms blocks, computes its voice value and depending on the\n"
"voice_threshold does the respective restoration. You can always get the\n"
"all-voiced or all-unvoiced cases by setting the voice_threshold to 0 or\n"
"1 respectively.\n\n";
// Read next buffers from the test files (signed 16-bit host-endian PCM
// format). audio_buffer has int16 samples, detection_buffer has float samples
// with range [-32768,32767], and reference_buffer has float samples with range
// [-1,1]. Return true iff all the buffers were filled completely.
bool ReadBuffers(FILE* in_file,
size_t audio_buffer_size,
int num_channels,
int16_t* audio_buffer,
FILE* detection_file,
size_t detection_buffer_size,
float* detection_buffer,
FILE* reference_file,
float* reference_buffer) {
std::unique_ptr<int16_t[]> tmpbuf;
int16_t* read_ptr = audio_buffer;
if (num_channels > 1) {
tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
read_ptr = tmpbuf.get();
if (fread(read_ptr, sizeof(*read_ptr), num_channels * audio_buffer_size,
in_file) != num_channels * audio_buffer_size) {
return false;
// De-interleave.
if (num_channels > 1) {
for (int i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < audio_buffer_size; ++j) {
audio_buffer[i * audio_buffer_size + j] =
read_ptr[i + j * num_channels];
if (detection_file) {
std::unique_ptr<int16_t[]> ibuf(new int16_t[detection_buffer_size]);
if (fread(ibuf.get(), sizeof(ibuf[0]), detection_buffer_size,
detection_file) != detection_buffer_size)
return false;
for (size_t i = 0; i < detection_buffer_size; ++i)
detection_buffer[i] = ibuf[i];
if (reference_file) {
std::unique_ptr<int16_t[]> ibuf(new int16_t[audio_buffer_size]);
if (fread(ibuf.get(), sizeof(ibuf[0]), audio_buffer_size, reference_file) !=
return false;
S16ToFloat(ibuf.get(), audio_buffer_size, reference_buffer);
return true;
// Write a number of samples to an open signed 16-bit host-endian PCM file.
static void WritePCM(FILE* f,
size_t num_samples,
int num_channels,
const float* buffer) {
std::unique_ptr<int16_t[]> ibuf(new int16_t[num_channels * num_samples]);
// Interleave.
for (int i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < num_samples; ++j) {
ibuf[i + j * num_channels] = FloatS16ToS16(buffer[i * num_samples + j]);
fwrite(ibuf.get(), sizeof(ibuf[0]), num_channels * num_samples, f);
// This application tests the transient suppression by providing a processed
// PCM file, which has to be listened to in order to evaluate the
// performance.
// It gets an audio file, and its voice gain information, and the suppressor
// process it giving the output file "suppressed_keystrokes.pcm".
void void_main() {
// TODO(aluebs): Remove all FileWrappers.
// Prepare the input file.
FILE* in_file = fopen(absl::GetFlag(FLAGS_in_file_name).c_str(), "rb");
ASSERT_TRUE(in_file != NULL);
// Prepare the detection file.
FILE* detection_file = NULL;
if (!absl::GetFlag(FLAGS_detection_file_name).empty()) {
detection_file =
fopen(absl::GetFlag(FLAGS_detection_file_name).c_str(), "rb");
// Prepare the reference file.
FILE* reference_file = NULL;
if (!absl::GetFlag(FLAGS_reference_file_name).empty()) {
reference_file =
fopen(absl::GetFlag(FLAGS_reference_file_name).c_str(), "rb");
// Prepare the output file.
std::string out_file_name = test::OutputPath() + "suppressed_keystrokes.pcm";
FILE* out_file = fopen(out_file_name.c_str(), "wb");
ASSERT_TRUE(out_file != NULL);
int detection_rate_hz = absl::GetFlag(FLAGS_detection_rate_hz);
if (detection_rate_hz == 0) {
detection_rate_hz = absl::GetFlag(FLAGS_sample_rate_hz);
Agc agc;
TransientSuppressorImpl suppressor;
suppressor.Initialize(absl::GetFlag(FLAGS_sample_rate_hz), detection_rate_hz,
const size_t audio_buffer_size = absl::GetFlag(FLAGS_chunk_size_ms) *
absl::GetFlag(FLAGS_sample_rate_hz) / 1000;
const size_t detection_buffer_size =
absl::GetFlag(FLAGS_chunk_size_ms) * detection_rate_hz / 1000;
// int16 and float variants of the same data.
std::unique_ptr<int16_t[]> audio_buffer_i(
new int16_t[absl::GetFlag(FLAGS_num_channels) * audio_buffer_size]);
std::unique_ptr<float[]> audio_buffer_f(
new float[absl::GetFlag(FLAGS_num_channels) * audio_buffer_size]);
std::unique_ptr<float[]> detection_buffer, reference_buffer;
if (detection_file)
detection_buffer.reset(new float[detection_buffer_size]);
if (reference_file)
reference_buffer.reset(new float[audio_buffer_size]);
while (ReadBuffers(
in_file, audio_buffer_size, absl::GetFlag(FLAGS_num_channels),
audio_buffer_i.get(), detection_file, detection_buffer_size,
detection_buffer.get(), reference_file, reference_buffer.get())) {
agc.Process(audio_buffer_i.get(), static_cast<int>(audio_buffer_size),
for (size_t i = 0;
i < absl::GetFlag(FLAGS_num_channels) * audio_buffer_size; ++i) {
audio_buffer_f[i] = audio_buffer_i[i];
ASSERT_EQ(0, suppressor.Suppress(
audio_buffer_f.get(), audio_buffer_size,
absl::GetFlag(FLAGS_num_channels), detection_buffer.get(),
detection_buffer_size, reference_buffer.get(),
audio_buffer_size, agc.voice_probability(), true))
<< "The transient suppressor could not suppress the frame";
// Write result to out file.
WritePCM(out_file, audio_buffer_size, absl::GetFlag(FLAGS_num_channels),
if (detection_file) {
if (reference_file) {
} // namespace webrtc
int main(int argc, char* argv[]) {
std::vector<char*> args = absl::ParseCommandLine(argc, argv);
if (args.size() != 1) {
printf("%s", webrtc::kUsage);
return 1;
RTC_CHECK_GT(absl::GetFlag(FLAGS_chunk_size_ms), 0);
RTC_CHECK_GT(absl::GetFlag(FLAGS_sample_rate_hz), 0);
RTC_CHECK_GT(absl::GetFlag(FLAGS_num_channels), 0);
return 0;