1 files changed, 96 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc
new file mode 100644
index 0000000000..73ce4ed3f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc
@@ -0,0 +1,96 @@
+// 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);
+}