| // 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/wav_file.h" |
| #include "modules/audio_processing/vad/voice_activity_detector.h" |
| #include "rtc_base/logging.h" |
| |
| ABSL_FLAG(std::string, i, "", "Input wav file"); |
| ABSL_FLAG(std::string, o_probs, "", "VAD probabilities output file"); |
| ABSL_FLAG(std::string, o_rms, "", "VAD output file"); |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| constexpr uint8_t kAudioFrameLengthMilliseconds = 10; |
| constexpr int kMaxSampleRate = 48000; |
| constexpr size_t kMaxFrameLen = |
| kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000; |
| |
| int main(int argc, char* argv[]) { |
| absl::ParseCommandLine(argc, argv); |
| const std::string input_file = absl::GetFlag(FLAGS_i); |
| const std::string output_probs_file = absl::GetFlag(FLAGS_o_probs); |
| const std::string output_file = absl::GetFlag(FLAGS_o_rms); |
| // 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_len = rtc::CheckedDivExact( |
| kAudioFrameLengthMilliseconds * wav_reader.sample_rate(), 1000); |
| if (audio_frame_len > 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_probs_file(output_probs_file, std::ofstream::binary); |
| std::ofstream out_rms_file(output_file, std::ofstream::binary); |
| |
| // Run VAD and write decisions. |
| VoiceActivityDetector vad; |
| std::array<int16_t, kMaxFrameLen> samples; |
| |
| while (true) { |
| // Process frame. |
| const auto read_samples = |
| wav_reader.ReadSamples(audio_frame_len, samples.data()); |
| if (read_samples < audio_frame_len) { |
| break; |
| } |
| vad.ProcessChunk(samples.data(), audio_frame_len, wav_reader.sample_rate()); |
| // Write output. |
| auto probs = vad.chunkwise_voice_probabilities(); |
| auto rms = vad.chunkwise_rms(); |
| RTC_CHECK_EQ(probs.size(), rms.size()); |
| RTC_CHECK_EQ(sizeof(double), 8); |
| |
| for (const auto& p : probs) { |
| out_probs_file.write(reinterpret_cast<const char*>(&p), 8); |
| } |
| for (const auto& r : rms) { |
| out_rms_file.write(reinterpret_cast<const char*>(&r), 8); |
| } |
| } |
| |
| out_probs_file.close(); |
| out_rms_file.close(); |
| return 0; |
| } |
| |
| } // namespace |
| } // namespace test |
| } // namespace webrtc |
| |
| int main(int argc, char* argv[]) { |
| return webrtc::test::main(argc, argv); |
| } |