1 files changed, 127 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
new file mode 100644
index 0000000000..1f24d9d370
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
@@ -0,0 +1,127 @@
+// 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 int16_t peak_level = *std::max_element(
+        samples.cbegin(), samples.cbegin() + 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);
+}