blob: d58b57e03cf2bee0d398ec43e10fffe0a86fea50 [file] [log] [blame]
// Copyright (c) 2018 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 <algorithm>
#include <array>
#include <cmath>
#include <fstream>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/wav_file.h"
#include "rtc_base/logging.h"
ABSL_FLAG(std::string, i, "", "Input wav file");
ABSL_FLAG(std::string, oc, "", "Config output file");
ABSL_FLAG(std::string, ol, "", "Levels output file");
ABSL_FLAG(float, a, 5.f, "Attack (ms)");
ABSL_FLAG(float, d, 20.f, "Decay (ms)");
ABSL_FLAG(int, f, 10, "Frame length (ms)");
namespace webrtc {
namespace test {
namespace {
constexpr int kMaxSampleRate = 48000;
constexpr uint8_t kMaxFrameLenMs = 30;
constexpr size_t kMaxFrameLen = kMaxFrameLenMs * kMaxSampleRate / 1000;
const double kOneDbReduction = DbToRatio(-1.0);
int main(int argc, char* argv[]) {
absl::ParseCommandLine(argc, argv);
// Check parameters.
if (absl::GetFlag(FLAGS_f) < 1 || absl::GetFlag(FLAGS_f) > kMaxFrameLenMs) {
RTC_LOG(LS_ERROR) << "Invalid frame length (min: 1, max: " << kMaxFrameLenMs
<< ")";
return 1;
}
if (absl::GetFlag(FLAGS_a) < 0 || absl::GetFlag(FLAGS_d) < 0) {
RTC_LOG(LS_ERROR) << "Attack and decay must be non-negative";
return 1;
}
// Open wav input file and check properties.
const std::string input_file = absl::GetFlag(FLAGS_i);
const std::string config_output_file = absl::GetFlag(FLAGS_oc);
const std::string levels_output_file = absl::GetFlag(FLAGS_ol);
WavReader wav_reader(input_file);
if (wav_reader.num_channels() != 1) {
RTC_LOG(LS_ERROR) << "Only mono wav files supported";
return 1;
}
if (wav_reader.sample_rate() > kMaxSampleRate) {
RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate
<< ")";
return 1;
}
// Map from milliseconds to samples.
const size_t audio_frame_length = rtc::CheckedDivExact(
absl::GetFlag(FLAGS_f) * wav_reader.sample_rate(), 1000);
auto time_const = [](double c) {
return std::pow(kOneDbReduction, absl::GetFlag(FLAGS_f) / c);
};
const float attack =
absl::GetFlag(FLAGS_a) == 0.0 ? 0.0 : time_const(absl::GetFlag(FLAGS_a));
const float decay =
absl::GetFlag(FLAGS_d) == 0.0 ? 0.0 : time_const(absl::GetFlag(FLAGS_d));
// Write config to file.
std::ofstream out_config(config_output_file);
out_config << "{"
"'frame_len_ms': "
<< absl::GetFlag(FLAGS_f)
<< ", "
"'attack_ms': "
<< absl::GetFlag(FLAGS_a)
<< ", "
"'decay_ms': "
<< absl::GetFlag(FLAGS_d) << "}\n";
out_config.close();
// Measure level frame-by-frame.
std::ofstream out_levels(levels_output_file, std::ofstream::binary);
std::array<int16_t, kMaxFrameLen> samples;
float level_prev = 0.f;
while (true) {
// Process frame.
const auto read_samples =
wav_reader.ReadSamples(audio_frame_length, samples.data());
if (read_samples < audio_frame_length)
break; // EOF.
// Frame peak level.
std::transform(samples.begin(), samples.begin() + audio_frame_length,
samples.begin(), [](int16_t s) { return std::abs(s); });
const auto* peak_level =
std::max_element(samples.begin(), samples.begin() + audio_frame_length);
const float level_curr = static_cast<float>(*peak_level) / 32768.f;
// Temporal smoothing.
auto smooth = [&level_prev, &level_curr](float c) {
return (1.0 - c) * level_curr + c * level_prev;
};
level_prev = smooth(level_curr > level_prev ? attack : decay);
// Write output.
out_levels.write(reinterpret_cast<const char*>(&level_prev), sizeof(float));
}
out_levels.close();
return 0;
}
} // namespace
} // namespace test
} // namespace webrtc
int main(int argc, char* argv[]) {
return webrtc::test::main(argc, argv);
}