blob: a55378570ff594d3bc55a51941894c9bf80d8145 [file] [log] [blame]
// Copyright (c) 2017 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 <array>
#include <fstream>
#include <memory>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "common_audio/vad/include/vad.h"
#include "common_audio/wav_file.h"
#include "rtc_base/logging.h"
ABSL_FLAG(std::string, i, "", "Input wav file");
ABSL_FLAG(std::string, o, "", "VAD output file");
namespace webrtc {
namespace test {
namespace {
// The allowed values are 10, 20 or 30 ms.
constexpr uint8_t kAudioFrameLengthMilliseconds = 30;
constexpr int kMaxSampleRate = 48000;
constexpr size_t kMaxFrameLen =
kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000;
constexpr uint8_t kBitmaskBuffSize = 8;
int main(int argc, char* argv[]) {
absl::ParseCommandLine(argc, argv);
const std::string input_file = absl::GetFlag(FLAGS_i);
const std::string output_file = absl::GetFlag(FLAGS_o);
// Open wav input file and check properties.
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;
}
const size_t audio_frame_length = rtc::CheckedDivExact(
kAudioFrameLengthMilliseconds * wav_reader.sample_rate(), 1000);
if (audio_frame_length > kMaxFrameLen) {
RTC_LOG(LS_ERROR) << "The frame size and/or the sample rate are too large.";
return 1;
}
// Create output file and write header.
std::ofstream out_file(output_file, std::ofstream::binary);
const char audio_frame_length_ms = kAudioFrameLengthMilliseconds;
out_file.write(&audio_frame_length_ms, 1); // Header.
// Run VAD and write decisions.
std::unique_ptr<Vad> vad = CreateVad(Vad::Aggressiveness::kVadNormal);
std::array<int16_t, kMaxFrameLen> samples;
char buff = 0; // Buffer to write one bit per frame.
uint8_t next = 0; // Points to the next bit to write in |buff|.
while (true) {
// Process frame.
const auto read_samples =
wav_reader.ReadSamples(audio_frame_length, samples.data());
if (read_samples < audio_frame_length)
break;
const auto is_speech = vad->VoiceActivity(
samples.data(), audio_frame_length, wav_reader.sample_rate());
// Write output.
buff = is_speech ? buff | (1 << next) : buff & ~(1 << next);
if (++next == kBitmaskBuffSize) {
out_file.write(&buff, 1); // Flush.
buff = 0; // Reset.
next = 0;
}
}
// Finalize.
char extra_bits = 0;
if (next > 0) {
extra_bits = kBitmaskBuffSize - next;
out_file.write(&buff, 1); // Flush.
}
out_file.write(&extra_bits, 1);
out_file.close();
return 0;
}
} // namespace
} // namespace test
} // namespace webrtc
int main(int argc, char* argv[]) {
return webrtc::test::main(argc, argv);
}