APM-QA anntator for sound level measurement

Bug: webrtc:7494
Change-Id: I6cdc282a1b3e0c0fbd8ef2e45d9b60af3b15a84b
Reviewed-on: https://webrtc-review.googlesource.com/40602
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21697}

modules/audio_processing/test/py_quality_assessment/BUILD.gn

@@ -146,6 +146,19 @@ rtc_executable("apm_vad") {
   ]
 }
 
+rtc_executable("sound_level") {
+  sources = [
+    "quality_assessment/sound_level.cc",
+  ]
+  deps = [
+    "../..",
+    "../../../..:webrtc_common",
+    "../../../../common_audio",
+    "../../../../rtc_base:rtc_base_approved",
+    "../../../../system_wrappers:metrics_default",
+  ]
+}
+
 copy("lib_unit_tests") {
   testonly = true
   sources = [

modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc

@@ -0,0 +1,124 @@
+// 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 "common_audio/wav_file.h"
+#include "rtc_base/flags.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kMaxSampleRate = 48000;
+constexpr uint8_t kMaxFrameLenMs = 30;
+constexpr size_t kMaxFrameLen = kMaxFrameLenMs * kMaxSampleRate / 1000;
+
+const double kOneDbReduction = std::pow(10.0, -1.0 / 20.0);
+
+DEFINE_string(i, "", "Input wav file");
+DEFINE_string(oc, "", "Config output file");
+DEFINE_string(ol, "", "Levels output file");
+DEFINE_float(a, 5.f, "Attack (ms)");
+DEFINE_float(d, 20.f, "Decay (ms)");
+DEFINE_int(f, 10, "Frame length (ms)");
+DEFINE_bool(help, false, "prints this message");
+
+int main(int argc, char* argv[]) {
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
+    rtc::FlagList::Print(nullptr, false);
+    return 1;
+  }
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
+
+  // Check parameters.
+  if (FLAG_f < 1 || FLAG_f > kMaxFrameLenMs) {
+    RTC_LOG(LS_ERROR) << "Invalid frame length (min: 1, max: " << kMaxFrameLenMs
+                      << ")";
+    return 1;
+  }
+  if (FLAG_a < 0 || FLAG_d < 0) {
+    RTC_LOG(LS_ERROR) << "Attack and decay must be non-negative";
+    return 1;
+  }
+
+  // Open wav input file and check properties.
+  WavReader wav_reader(FLAG_i);
+  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(FLAG_f * wav_reader.sample_rate(), 1000);
+  auto time_const = [](double c) {
+    return std::pow(kOneDbReduction, FLAG_f / c);
+  };
+  const float attack = FLAG_a == 0.0 ? 0.0 : time_const(FLAG_a);
+  const float decay = FLAG_d == 0.0 ? 0.0 : time_const(FLAG_d);
+
+  // Write config to file.
+  std::ofstream out_config(FLAG_oc);
+  out_config << "{"
+             << "'frame_len_ms': " << FLAG_f << ", "
+             << "'attack_ms': " << FLAG_a << ", "
+             << "'decay_ms': " << FLAG_d << "}\n";
+  out_config.close();
+
+  // Measure level frame-by-frame.
+  std::ofstream out_levels(FLAG_ol, 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 auto* peak_level =
+        std::max_element(samples.begin(), samples.begin() + 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);
+}