1 files changed, 121 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
new file mode 100644
index 0000000000..1995b24913
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
@@ -0,0 +1,121 @@
+/*
+ *  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/fixed_digital_level_estimator.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kInitialFilterStateLevel = 0.0f;
+
+// Instant attack.
+constexpr float kAttackFilterConstant = 0.0f;
+
+// Limiter decay constant.
+// Computed as `10 ** (-1/20 * subframe_duration / kDecayMs)` where:
+// - `subframe_duration` is `kFrameDurationMs / kSubFramesInFrame`;
+// - `kDecayMs` is defined in agc2_testing_common.h.
+constexpr float kDecayFilterConstant = 0.9971259f;
+
+}  // namespace
+
+FixedDigitalLevelEstimator::FixedDigitalLevelEstimator(
+    int sample_rate_hz,
+    ApmDataDumper* apm_data_dumper)
+    : apm_data_dumper_(apm_data_dumper),
+      filter_state_level_(kInitialFilterStateLevel) {
+  SetSampleRate(sample_rate_hz);
+  CheckParameterCombination();
+  RTC_DCHECK(apm_data_dumper_);
+  apm_data_dumper_->DumpRaw("agc2_level_estimator_samplerate", sample_rate_hz);
+}
+
+void FixedDigitalLevelEstimator::CheckParameterCombination() {
+  RTC_DCHECK_GT(samples_in_frame_, 0);
+  RTC_DCHECK_LE(kSubFramesInFrame, samples_in_frame_);
+  RTC_DCHECK_EQ(samples_in_frame_ % kSubFramesInFrame, 0);
+  RTC_DCHECK_GT(samples_in_sub_frame_, 1);
+}
+
+std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
+    const AudioFrameView<const float>& float_frame) {
+  RTC_DCHECK_GT(float_frame.num_channels(), 0);
+  RTC_DCHECK_EQ(float_frame.samples_per_channel(), samples_in_frame_);
+
+  // Compute max envelope without smoothing.
+  std::array<float, kSubFramesInFrame> envelope{};
+  for (int channel_idx = 0; channel_idx < float_frame.num_channels();
+       ++channel_idx) {
+    const auto channel = float_frame.channel(channel_idx);
+    for (int sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
+      for (int sample_in_sub_frame = 0;
+           sample_in_sub_frame < samples_in_sub_frame_; ++sample_in_sub_frame) {
+        envelope[sub_frame] =
+            std::max(envelope[sub_frame],
+                     std::abs(channel[sub_frame * samples_in_sub_frame_ +
+                                      sample_in_sub_frame]));
+      }
+    }
+  }
+
+  // Make sure envelope increases happen one step earlier so that the
+  // corresponding *gain decrease* doesn't miss a sudden signal
+  // increase due to interpolation.
+  for (int sub_frame = 0; sub_frame < kSubFramesInFrame - 1; ++sub_frame) {
+    if (envelope[sub_frame] < envelope[sub_frame + 1]) {
+      envelope[sub_frame] = envelope[sub_frame + 1];
+    }
+  }
+
+  // Add attack / decay smoothing.
+  for (int sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
+    const float envelope_value = envelope[sub_frame];
+    if (envelope_value > filter_state_level_) {
+      envelope[sub_frame] = envelope_value * (1 - kAttackFilterConstant) +
+                            filter_state_level_ * kAttackFilterConstant;
+    } else {
+      envelope[sub_frame] = envelope_value * (1 - kDecayFilterConstant) +
+                            filter_state_level_ * kDecayFilterConstant;
+    }
+    filter_state_level_ = envelope[sub_frame];
+
+    // Dump data for debug.
+    RTC_DCHECK(apm_data_dumper_);
+    const auto channel = float_frame.channel(0);
+    apm_data_dumper_->DumpRaw("agc2_level_estimator_samples",
+                              samples_in_sub_frame_,
+                              &channel[sub_frame * samples_in_sub_frame_]);
+    apm_data_dumper_->DumpRaw("agc2_level_estimator_level",
+                              envelope[sub_frame]);
+  }
+
+  return envelope;
+}
+
+void FixedDigitalLevelEstimator::SetSampleRate(int sample_rate_hz) {
+  samples_in_frame_ =
+      rtc::CheckedDivExact(sample_rate_hz * kFrameDurationMs, 1000);
+  samples_in_sub_frame_ =
+      rtc::CheckedDivExact(samples_in_frame_, kSubFramesInFrame);
+  CheckParameterCombination();
+}
+
+void FixedDigitalLevelEstimator::Reset() {
+  filter_state_level_ = kInitialFilterStateLevel;
+}
+
+}  // namespace webrtc