1 files changed, 155 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter.cc b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.cc
new file mode 100644
index 0000000000..7a1e2202be
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.cc
@@ -0,0 +1,155 @@
+/*
+ *  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 "modules/audio_processing/agc2/limiter.h"
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+// This constant affects the way scaling factors are interpolated for the first
+// sub-frame of a frame. Only in the case in which the first sub-frame has an
+// estimated level which is greater than the that of the previous analyzed
+// sub-frame, linear interpolation is replaced with a power function which
+// reduces the chances of over-shooting (and hence saturation), however reducing
+// the fixed gain effectiveness.
+constexpr float kAttackFirstSubframeInterpolationPower = 8.0f;
+
+void InterpolateFirstSubframe(float last_factor,
+                              float current_factor,
+                              rtc::ArrayView<float> subframe) {
+  const int n = rtc::dchecked_cast<int>(subframe.size());
+  constexpr float p = kAttackFirstSubframeInterpolationPower;
+  for (int i = 0; i < n; ++i) {
+    subframe[i] = std::pow(1.f - i / n, p) * (last_factor - current_factor) +
+                  current_factor;
+  }
+}
+
+void ComputePerSampleSubframeFactors(
+    const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
+    int samples_per_channel,
+    rtc::ArrayView<float> per_sample_scaling_factors) {
+  const int num_subframes = scaling_factors.size() - 1;
+  const int subframe_size =
+      rtc::CheckedDivExact(samples_per_channel, num_subframes);
+
+  // Handle first sub-frame differently in case of attack.
+  const bool is_attack = scaling_factors[0] > scaling_factors[1];
+  if (is_attack) {
+    InterpolateFirstSubframe(
+        scaling_factors[0], scaling_factors[1],
+        rtc::ArrayView<float>(
+            per_sample_scaling_factors.subview(0, subframe_size)));
+  }
+
+  for (int i = is_attack ? 1 : 0; i < num_subframes; ++i) {
+    const int subframe_start = i * subframe_size;
+    const float scaling_start = scaling_factors[i];
+    const float scaling_end = scaling_factors[i + 1];
+    const float scaling_diff = (scaling_end - scaling_start) / subframe_size;
+    for (int j = 0; j < subframe_size; ++j) {
+      per_sample_scaling_factors[subframe_start + j] =
+          scaling_start + scaling_diff * j;
+    }
+  }
+}
+
+void ScaleSamples(rtc::ArrayView<const float> per_sample_scaling_factors,
+                  AudioFrameView<float> signal) {
+  const int samples_per_channel = signal.samples_per_channel();
+  RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size());
+  for (int i = 0; i < signal.num_channels(); ++i) {
+    rtc::ArrayView<float> channel = signal.channel(i);
+    for (int j = 0; j < samples_per_channel; ++j) {
+      channel[j] = rtc::SafeClamp(channel[j] * per_sample_scaling_factors[j],
+                                  kMinFloatS16Value, kMaxFloatS16Value);
+    }
+  }
+}
+
+void CheckLimiterSampleRate(int sample_rate_hz) {
+  // Check that per_sample_scaling_factors_ is large enough.
+  RTC_DCHECK_LE(sample_rate_hz,
+                kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
+}
+
+}  // namespace
+
+Limiter::Limiter(int sample_rate_hz,
+                 ApmDataDumper* apm_data_dumper,
+                 absl::string_view histogram_name)
+    : interp_gain_curve_(apm_data_dumper, histogram_name),
+      level_estimator_(sample_rate_hz, apm_data_dumper),
+      apm_data_dumper_(apm_data_dumper) {
+  CheckLimiterSampleRate(sample_rate_hz);
+}
+
+Limiter::~Limiter() = default;
+
+void Limiter::Process(AudioFrameView<float> signal) {
+  const std::array<float, kSubFramesInFrame> level_estimate =
+      level_estimator_.ComputeLevel(signal);
+
+  RTC_DCHECK_EQ(level_estimate.size() + 1, scaling_factors_.size());
+  scaling_factors_[0] = last_scaling_factor_;
+  std::transform(level_estimate.begin(), level_estimate.end(),
+                 scaling_factors_.begin() + 1, [this](float x) {
+                   return interp_gain_curve_.LookUpGainToApply(x);
+                 });
+
+  const int samples_per_channel = signal.samples_per_channel();
+  RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
+
+  auto per_sample_scaling_factors = rtc::ArrayView<float>(
+      &per_sample_scaling_factors_[0], samples_per_channel);
+  ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel,
+                                  per_sample_scaling_factors);
+  ScaleSamples(per_sample_scaling_factors, signal);
+
+  last_scaling_factor_ = scaling_factors_.back();
+
+  // Dump data for debug.
+  apm_data_dumper_->DumpRaw("agc2_limiter_last_scaling_factor",
+                            last_scaling_factor_);
+  apm_data_dumper_->DumpRaw(
+      "agc2_limiter_region",
+      static_cast<int>(interp_gain_curve_.get_stats().region));
+}
+
+InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {
+  return interp_gain_curve_.get_stats();
+}
+
+void Limiter::SetSampleRate(int sample_rate_hz) {
+  CheckLimiterSampleRate(sample_rate_hz);
+  level_estimator_.SetSampleRate(sample_rate_hz);
+}
+
+void Limiter::Reset() {
+  level_estimator_.Reset();
+}
+
+float Limiter::LastAudioLevel() const {
+  return level_estimator_.LastAudioLevel();
+}
+
+}  // namespace webrtc