1 files changed, 103 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
new file mode 100644
index 0000000000..b47f6221cb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
@@ -0,0 +1,103 @@
+// 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 "common_audio/vad/include/vad.h"
+
+#include <array>
+#include <fstream>
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.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);
+}