Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

58 files changed, 8058 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
new file mode 100644
index 0000000000..d709eb3699
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -0,0 +1,334 @@
+# 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.
+
+import("../../../../webrtc.gni")
+
+rtc_library("rnn_vad") {
+  visibility = [ "../*" ]
+  sources = [
+    "features_extraction.cc",
+    "features_extraction.h",
+    "rnn.cc",
+    "rnn.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && target_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":rnn_vad_common",
+    ":rnn_vad_layers",
+    ":rnn_vad_lp_residual",
+    ":rnn_vad_pitch",
+    ":rnn_vad_sequence_buffer",
+    ":rnn_vad_spectral_features",
+    "..:biquad_filter",
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+    "../../../../rtc_base:safe_conversions",
+    "//third_party/rnnoise:rnn_vad",
+  ]
+}
+
+rtc_library("rnn_vad_auto_correlation") {
+  sources = [
+    "auto_correlation.cc",
+    "auto_correlation.h",
+  ]
+  deps = [
+    ":rnn_vad_common",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../utility:pffft_wrapper",
+  ]
+}
+
+rtc_source_set("rnn_vad_common") {
+  # TODO(alessiob): Make this target visibility private.
+  visibility = [
+    ":*",
+    "..:vad_wrapper",
+  ]
+  sources = [ "common.h" ]
+  deps = [
+    "../../../../rtc_base/system:arch",
+    "../../../../system_wrappers",
+  ]
+}
+
+rtc_library("rnn_vad_lp_residual") {
+  sources = [
+    "lp_residual.cc",
+    "lp_residual.h",
+  ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+  ]
+}
+
+rtc_source_set("rnn_vad_layers") {
+  sources = [
+    "rnn_fc.cc",
+    "rnn_fc.h",
+    "rnn_gru.cc",
+    "rnn_gru.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && current_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":rnn_vad_common",
+    ":vector_math",
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../api:function_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_conversions",
+    "//third_party/rnnoise:rnn_vad",
+  ]
+  if (current_cpu == "x86" || current_cpu == "x64") {
+    deps += [ ":vector_math_avx2" ]
+  }
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_source_set("vector_math") {
+  sources = [ "vector_math.h" ]
+  deps = [
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_conversions",
+    "../../../../rtc_base/system:arch",
+  ]
+}
+
+if (current_cpu == "x86" || current_cpu == "x64") {
+  rtc_library("vector_math_avx2") {
+    sources = [ "vector_math_avx2.cc" ]
+    if (is_win && !build_with_mozilla) {
+      cflags = [ "/arch:AVX2" ]
+    } else {
+      cflags = [
+        "-mavx2",
+        "-mfma",
+      ]
+    }
+    deps = [
+      ":vector_math",
+      "../../../../api:array_view",
+      "../../../../rtc_base:checks",
+      "../../../../rtc_base:safe_conversions",
+    ]
+  }
+}
+
+rtc_library("rnn_vad_pitch") {
+  sources = [
+    "pitch_search.cc",
+    "pitch_search.h",
+    "pitch_search_internal.cc",
+    "pitch_search_internal.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && current_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":rnn_vad_auto_correlation",
+    ":rnn_vad_common",
+    ":vector_math",
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:gtest_prod",
+    "../../../../rtc_base:safe_compare",
+    "../../../../rtc_base:safe_conversions",
+    "../../../../rtc_base/system:arch",
+  ]
+  if (current_cpu == "x86" || current_cpu == "x64") {
+    deps += [ ":vector_math_avx2" ]
+  }
+}
+
+rtc_source_set("rnn_vad_ring_buffer") {
+  sources = [ "ring_buffer.h" ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+  ]
+}
+
+rtc_source_set("rnn_vad_sequence_buffer") {
+  sources = [ "sequence_buffer.h" ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+  ]
+}
+
+rtc_library("rnn_vad_spectral_features") {
+  sources = [
+    "spectral_features.cc",
+    "spectral_features.h",
+    "spectral_features_internal.cc",
+    "spectral_features_internal.h",
+  ]
+  deps = [
+    ":rnn_vad_common",
+    ":rnn_vad_ring_buffer",
+    ":rnn_vad_symmetric_matrix_buffer",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+    "../../utility:pffft_wrapper",
+  ]
+}
+
+rtc_source_set("rnn_vad_symmetric_matrix_buffer") {
+  sources = [ "symmetric_matrix_buffer.h" ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("test_utils") {
+    testonly = true
+    sources = [
+      "test_utils.cc",
+      "test_utils.h",
+    ]
+    deps = [
+      ":rnn_vad",
+      ":rnn_vad_common",
+      "../../../../api:array_view",
+      "../../../../api:scoped_refptr",
+      "../../../../rtc_base:checks",
+      "../../../../rtc_base:safe_compare",
+      "../../../../test:fileutils",
+      "../../../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  unittest_resources = [
+    "../../../../resources/audio_processing/agc2/rnn_vad/band_energies.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_buf_24k.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_lp_res.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_search_int.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/samples.pcm",
+    "../../../../resources/audio_processing/agc2/rnn_vad/vad_prob.dat",
+  ]
+
+  if (is_ios) {
+    bundle_data("unittests_bundle_data") {
+      testonly = true
+      sources = unittest_resources
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+  }
+
+  rtc_library("unittests") {
+    testonly = true
+    sources = [
+      "auto_correlation_unittest.cc",
+      "features_extraction_unittest.cc",
+      "lp_residual_unittest.cc",
+      "pitch_search_internal_unittest.cc",
+      "pitch_search_unittest.cc",
+      "ring_buffer_unittest.cc",
+      "rnn_fc_unittest.cc",
+      "rnn_gru_unittest.cc",
+      "rnn_unittest.cc",
+      "rnn_vad_unittest.cc",
+      "sequence_buffer_unittest.cc",
+      "spectral_features_internal_unittest.cc",
+      "spectral_features_unittest.cc",
+      "symmetric_matrix_buffer_unittest.cc",
+      "vector_math_unittest.cc",
+    ]
+
+    defines = []
+    if (rtc_build_with_neon && current_cpu != "arm64") {
+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags = [ "-mfpu=neon" ]
+    }
+
+    deps = [
+      ":rnn_vad",
+      ":rnn_vad_auto_correlation",
+      ":rnn_vad_common",
+      ":rnn_vad_layers",
+      ":rnn_vad_lp_residual",
+      ":rnn_vad_pitch",
+      ":rnn_vad_ring_buffer",
+      ":rnn_vad_sequence_buffer",
+      ":rnn_vad_spectral_features",
+      ":rnn_vad_symmetric_matrix_buffer",
+      ":test_utils",
+      ":vector_math",
+      "..:cpu_features",
+      "../..:audioproc_test_utils",
+      "../../../../api:array_view",
+      "../../../../common_audio/",
+      "../../../../rtc_base:checks",
+      "../../../../rtc_base:logging",
+      "../../../../rtc_base:safe_compare",
+      "../../../../rtc_base:safe_conversions",
+      "../../../../rtc_base:stringutils",
+      "../../../../rtc_base/system:arch",
+      "../../../../test:test_support",
+      "../../utility:pffft_wrapper",
+      "//third_party/rnnoise:rnn_vad",
+    ]
+    if (current_cpu == "x86" || current_cpu == "x64") {
+      deps += [ ":vector_math_avx2" ]
+    }
+    absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+    data = unittest_resources
+    if (is_ios) {
+      deps += [ ":unittests_bundle_data" ]
+    }
+  }
+
+  if (!build_with_chromium) {
+    rtc_executable("rnn_vad_tool") {
+      testonly = true
+      sources = [ "rnn_vad_tool.cc" ]
+      deps = [
+        ":rnn_vad",
+        ":rnn_vad_common",
+        "..:cpu_features",
+        "../../../../api:array_view",
+        "../../../../common_audio",
+        "../../../../rtc_base:logging",
+        "../../../../rtc_base:safe_compare",
+        "../../../../test:test_support",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+      ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/DEPS b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/DEPS
new file mode 100644
index 0000000000..773c2d7edd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+  "+third_party/rnnoise",
+]
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
new file mode 100644
index 0000000000..3ddeec8dba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2019 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/rnn_vad/auto_correlation.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kAutoCorrelationFftOrder = 9;  // Length-512 FFT.
+static_assert(1 << kAutoCorrelationFftOrder >
+                  kNumLags12kHz + kBufSize12kHz - kMaxPitch12kHz,
+              "");
+
+}  // namespace
+
+AutoCorrelationCalculator::AutoCorrelationCalculator()
+    : fft_(1 << kAutoCorrelationFftOrder, Pffft::FftType::kReal),
+      tmp_(fft_.CreateBuffer()),
+      X_(fft_.CreateBuffer()),
+      H_(fft_.CreateBuffer()) {}
+
+AutoCorrelationCalculator::~AutoCorrelationCalculator() = default;
+
+// The auto-correlations coefficients are computed as follows:
+// |.........|...........|  <- pitch buffer
+//           [ x (fixed) ]
+// [   y_0   ]
+//         [ y_{m-1} ]
+// x and y are sub-array of equal length; x is never moved, whereas y slides.
+// The cross-correlation between y_0 and x corresponds to the auto-correlation
+// for the maximum pitch period. Hence, the first value in `auto_corr` has an
+// inverted lag equal to 0 that corresponds to a lag equal to the maximum
+// pitch period.
+void AutoCorrelationCalculator::ComputeOnPitchBuffer(
+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buf,
+    rtc::ArrayView<float, kNumLags12kHz> auto_corr) {
+  RTC_DCHECK_LT(auto_corr.size(), kMaxPitch12kHz);
+  RTC_DCHECK_GT(pitch_buf.size(), kMaxPitch12kHz);
+  constexpr int kFftFrameSize = 1 << kAutoCorrelationFftOrder;
+  constexpr int kConvolutionLength = kBufSize12kHz - kMaxPitch12kHz;
+  static_assert(kConvolutionLength == kFrameSize20ms12kHz,
+                "Mismatch between pitch buffer size, frame size and maximum "
+                "pitch period.");
+  static_assert(kFftFrameSize > kNumLags12kHz + kConvolutionLength,
+                "The FFT length is not sufficiently big to avoid cyclic "
+                "convolution errors.");
+  auto tmp = tmp_->GetView();
+
+  // Compute the FFT for the reversed reference frame - i.e.,
+  // pitch_buf[-kConvolutionLength:].
+  std::reverse_copy(pitch_buf.end() - kConvolutionLength, pitch_buf.end(),
+                    tmp.begin());
+  std::fill(tmp.begin() + kConvolutionLength, tmp.end(), 0.f);
+  fft_.ForwardTransform(*tmp_, H_.get(), /*ordered=*/false);
+
+  // Compute the FFT for the sliding frames chunk. The sliding frames are
+  // defined as pitch_buf[i:i+kConvolutionLength] where i in
+  // [0, kNumLags12kHz). The chunk includes all of them, hence it is
+  // defined as pitch_buf[:kNumLags12kHz+kConvolutionLength].
+  std::copy(pitch_buf.begin(),
+            pitch_buf.begin() + kConvolutionLength + kNumLags12kHz,
+            tmp.begin());
+  std::fill(tmp.begin() + kNumLags12kHz + kConvolutionLength, tmp.end(), 0.f);
+  fft_.ForwardTransform(*tmp_, X_.get(), /*ordered=*/false);
+
+  // Convolve in the frequency domain.
+  constexpr float kScalingFactor = 1.f / static_cast<float>(kFftFrameSize);
+  std::fill(tmp.begin(), tmp.end(), 0.f);
+  fft_.FrequencyDomainConvolve(*X_, *H_, tmp_.get(), kScalingFactor);
+  fft_.BackwardTransform(*tmp_, tmp_.get(), /*ordered=*/false);
+
+  // Extract the auto-correlation coefficients.
+  std::copy(tmp.begin() + kConvolutionLength - 1,
+            tmp.begin() + kConvolutionLength + kNumLags12kHz - 1,
+            auto_corr.begin());
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
new file mode 100644
index 0000000000..1ae5054567
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2019 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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_AUTO_CORRELATION_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_AUTO_CORRELATION_H_
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Class to compute the auto correlation on the pitch buffer for a target pitch
+// interval.
+class AutoCorrelationCalculator {
+ public:
+  AutoCorrelationCalculator();
+  AutoCorrelationCalculator(const AutoCorrelationCalculator&) = delete;
+  AutoCorrelationCalculator& operator=(const AutoCorrelationCalculator&) =
+      delete;
+  ~AutoCorrelationCalculator();
+
+  // Computes the auto-correlation coefficients for a target pitch interval.
+  // `auto_corr` indexes are inverted lags.
+  void ComputeOnPitchBuffer(
+      rtc::ArrayView<const float, kBufSize12kHz> pitch_buf,
+      rtc::ArrayView<float, kNumLags12kHz> auto_corr);
+
+ private:
+  Pffft fft_;
+  std::unique_ptr<Pffft::FloatBuffer> tmp_;
+  std::unique_ptr<Pffft::FloatBuffer> X_;
+  std::unique_ptr<Pffft::FloatBuffer> H_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_AUTO_CORRELATION_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation_unittest.cc
new file mode 100644
index 0000000000..76001ed7b7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation_unittest.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2019 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/rnn_vad/auto_correlation.h"
+
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Checks that the auto correlation function produces output within tolerance
+// given test input data.
+TEST(RnnVadTest, PitchBufferAutoCorrelationWithinTolerance) {
+  PitchTestData test_data;
+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
+  Decimate2x(test_data.PitchBuffer24kHzView(), pitch_buf_decimated);
+  std::array<float, kNumLags12kHz> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    AutoCorrelationCalculator auto_corr_calculator;
+    auto_corr_calculator.ComputeOnPitchBuffer(pitch_buf_decimated,
+                                              computed_output);
+  }
+  auto auto_corr_view = test_data.AutoCorrelation12kHzView();
+  ExpectNearAbsolute({auto_corr_view.data(), auto_corr_view.size()},
+                     computed_output, 3e-3f);
+}
+
+// Checks that the auto correlation function computes the right thing for a
+// simple use case.
+TEST(RnnVadTest, CheckAutoCorrelationOnConstantPitchBuffer) {
+  // Create constant signal with no pitch.
+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
+  std::fill(pitch_buf_decimated.begin(), pitch_buf_decimated.end(), 1.f);
+  std::array<float, kNumLags12kHz> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    AutoCorrelationCalculator auto_corr_calculator;
+    auto_corr_calculator.ComputeOnPitchBuffer(pitch_buf_decimated,
+                                              computed_output);
+  }
+  // The expected output is a vector filled with the same expected
+  // auto-correlation value. The latter equals the length of a 20 ms frame.
+  constexpr int kFrameSize20ms12kHz = kFrameSize20ms24kHz / 2;
+  std::array<float, kNumLags12kHz> expected_output;
+  std::fill(expected_output.begin(), expected_output.end(),
+            static_cast<float>(kFrameSize20ms12kHz));
+  ExpectNearAbsolute(expected_output, computed_output, 4e-5f);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/common.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/common.h
new file mode 100644
index 0000000000..c099373200
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/common.h
@@ -0,0 +1,77 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+namespace rnn_vad {
+
+constexpr double kPi = 3.14159265358979323846;
+
+constexpr int kSampleRate24kHz = 24000;
+constexpr int kFrameSize10ms24kHz = kSampleRate24kHz / 100;
+constexpr int kFrameSize20ms24kHz = kFrameSize10ms24kHz * 2;
+
+// Pitch buffer.
+constexpr int kMinPitch24kHz = kSampleRate24kHz / 800;   // 0.00125 s.
+constexpr int kMaxPitch24kHz = kSampleRate24kHz / 62.5;  // 0.016 s.
+constexpr int kBufSize24kHz = kMaxPitch24kHz + kFrameSize20ms24kHz;
+static_assert((kBufSize24kHz & 1) == 0, "The buffer size must be even.");
+
+// 24 kHz analysis.
+// Define a higher minimum pitch period for the initial search. This is used to
+// avoid searching for very short periods, for which a refinement step is
+// responsible.
+constexpr int kInitialMinPitch24kHz = 3 * kMinPitch24kHz;
+static_assert(kMinPitch24kHz < kInitialMinPitch24kHz, "");
+static_assert(kInitialMinPitch24kHz < kMaxPitch24kHz, "");
+static_assert(kMaxPitch24kHz > kInitialMinPitch24kHz, "");
+// Number of (inverted) lags during the initial pitch search phase at 24 kHz.
+constexpr int kInitialNumLags24kHz = kMaxPitch24kHz - kInitialMinPitch24kHz;
+// Number of (inverted) lags during the pitch search refinement phase at 24 kHz.
+constexpr int kRefineNumLags24kHz = kMaxPitch24kHz + 1;
+static_assert(
+    kRefineNumLags24kHz > kInitialNumLags24kHz,
+    "The refinement step must search the pitch in an extended pitch range.");
+
+// 12 kHz analysis.
+constexpr int kSampleRate12kHz = 12000;
+constexpr int kFrameSize10ms12kHz = kSampleRate12kHz / 100;
+constexpr int kFrameSize20ms12kHz = kFrameSize10ms12kHz * 2;
+constexpr int kBufSize12kHz = kBufSize24kHz / 2;
+constexpr int kInitialMinPitch12kHz = kInitialMinPitch24kHz / 2;
+constexpr int kMaxPitch12kHz = kMaxPitch24kHz / 2;
+static_assert(kMaxPitch12kHz > kInitialMinPitch12kHz, "");
+// The inverted lags for the pitch interval [`kInitialMinPitch12kHz`,
+// `kMaxPitch12kHz`] are in the range [0, `kNumLags12kHz`].
+constexpr int kNumLags12kHz = kMaxPitch12kHz - kInitialMinPitch12kHz;
+
+// 48 kHz constants.
+constexpr int kMinPitch48kHz = kMinPitch24kHz * 2;
+constexpr int kMaxPitch48kHz = kMaxPitch24kHz * 2;
+
+// Spectral features.
+constexpr int kNumBands = 22;
+constexpr int kNumLowerBands = 6;
+static_assert((0 < kNumLowerBands) && (kNumLowerBands < kNumBands), "");
+constexpr int kCepstralCoeffsHistorySize = 8;
+static_assert(kCepstralCoeffsHistorySize > 2,
+              "The history size must at least be 3 to compute first and second "
+              "derivatives.");
+
+constexpr int kFeatureVectorSize = 42;
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
new file mode 100644
index 0000000000..502023428d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
@@ -0,0 +1,90 @@
+/*
+ *  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/rnn_vad/features_extraction.h"
+
+#include <array>
+
+#include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Computed as `scipy.signal.butter(N=2, Wn=60/24000, btype='highpass')`.
+constexpr BiQuadFilter::Config kHpfConfig24k{
+    {0.99446179f, -1.98892358f, 0.99446179f},
+    {-1.98889291f, 0.98895425f}};
+
+}  // namespace
+
+FeaturesExtractor::FeaturesExtractor(const AvailableCpuFeatures& cpu_features)
+    : use_high_pass_filter_(false),
+      hpf_(kHpfConfig24k),
+      pitch_buf_24kHz_(),
+      pitch_buf_24kHz_view_(pitch_buf_24kHz_.GetBufferView()),
+      lp_residual_(kBufSize24kHz),
+      lp_residual_view_(lp_residual_.data(), kBufSize24kHz),
+      pitch_estimator_(cpu_features),
+      reference_frame_view_(pitch_buf_24kHz_.GetMostRecentValuesView()) {
+  RTC_DCHECK_EQ(kBufSize24kHz, lp_residual_.size());
+  Reset();
+}
+
+FeaturesExtractor::~FeaturesExtractor() = default;
+
+void FeaturesExtractor::Reset() {
+  pitch_buf_24kHz_.Reset();
+  spectral_features_extractor_.Reset();
+  if (use_high_pass_filter_) {
+    hpf_.Reset();
+  }
+}
+
+bool FeaturesExtractor::CheckSilenceComputeFeatures(
+    rtc::ArrayView<const float, kFrameSize10ms24kHz> samples,
+    rtc::ArrayView<float, kFeatureVectorSize> feature_vector) {
+  // Pre-processing.
+  if (use_high_pass_filter_) {
+    std::array<float, kFrameSize10ms24kHz> samples_filtered;
+    hpf_.Process(samples, samples_filtered);
+    // Feed buffer with the pre-processed version of `samples`.
+    pitch_buf_24kHz_.Push(samples_filtered);
+  } else {
+    // Feed buffer with `samples`.
+    pitch_buf_24kHz_.Push(samples);
+  }
+  // Extract the LP residual.
+  float lpc_coeffs[kNumLpcCoefficients];
+  ComputeAndPostProcessLpcCoefficients(pitch_buf_24kHz_view_, lpc_coeffs);
+  ComputeLpResidual(lpc_coeffs, pitch_buf_24kHz_view_, lp_residual_view_);
+  // Estimate pitch on the LP-residual and write the normalized pitch period
+  // into the output vector (normalization based on training data stats).
+  pitch_period_48kHz_ = pitch_estimator_.Estimate(lp_residual_view_);
+  feature_vector[kFeatureVectorSize - 2] = 0.01f * (pitch_period_48kHz_ - 300);
+  // Extract lagged frames (according to the estimated pitch period).
+  RTC_DCHECK_LE(pitch_period_48kHz_ / 2, kMaxPitch24kHz);
+  auto lagged_frame = pitch_buf_24kHz_view_.subview(
+      kMaxPitch24kHz - pitch_period_48kHz_ / 2, kFrameSize20ms24kHz);
+  // Analyze reference and lagged frames checking if silence has been detected
+  // and write the feature vector.
+  return spectral_features_extractor_.CheckSilenceComputeFeatures(
+      reference_frame_view_, {lagged_frame.data(), kFrameSize20ms24kHz},
+      {feature_vector.data() + kNumLowerBands, kNumBands - kNumLowerBands},
+      {feature_vector.data(), kNumLowerBands},
+      {feature_vector.data() + kNumBands, kNumLowerBands},
+      {feature_vector.data() + kNumBands + kNumLowerBands, kNumLowerBands},
+      {feature_vector.data() + kNumBands + 2 * kNumLowerBands, kNumLowerBands},
+      &feature_vector[kFeatureVectorSize - 1]);
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
new file mode 100644
index 0000000000..d47a85bfb0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
@@ -0,0 +1,61 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/biquad_filter.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
+#include "modules/audio_processing/agc2/rnn_vad/sequence_buffer.h"
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Feature extractor to feed the VAD RNN.
+class FeaturesExtractor {
+ public:
+  explicit FeaturesExtractor(const AvailableCpuFeatures& cpu_features);
+  FeaturesExtractor(const FeaturesExtractor&) = delete;
+  FeaturesExtractor& operator=(const FeaturesExtractor&) = delete;
+  ~FeaturesExtractor();
+  void Reset();
+  // Analyzes the samples, computes the feature vector and returns true if
+  // silence is detected (false if not). When silence is detected,
+  // `feature_vector` is partially written and therefore must not be used to
+  // feed the VAD RNN.
+  bool CheckSilenceComputeFeatures(
+      rtc::ArrayView<const float, kFrameSize10ms24kHz> samples,
+      rtc::ArrayView<float, kFeatureVectorSize> feature_vector);
+
+ private:
+  const bool use_high_pass_filter_;
+  // TODO(bugs.webrtc.org/7494): Remove HPF depending on how AGC2 is used in APM
+  // and on whether an HPF is already used as pre-processing step in APM.
+  BiQuadFilter hpf_;
+  SequenceBuffer<float, kBufSize24kHz, kFrameSize10ms24kHz, kFrameSize20ms24kHz>
+      pitch_buf_24kHz_;
+  rtc::ArrayView<const float, kBufSize24kHz> pitch_buf_24kHz_view_;
+  std::vector<float> lp_residual_;
+  rtc::ArrayView<float, kBufSize24kHz> lp_residual_view_;
+  PitchEstimator pitch_estimator_;
+  rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame_view_;
+  SpectralFeaturesExtractor spectral_features_extractor_;
+  int pitch_period_48kHz_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
new file mode 100644
index 0000000000..96f956adfe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
@@ -0,0 +1,103 @@
+/*
+ *  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/rnn_vad/features_extraction.h"
+
+#include <cmath>
+#include <vector>
+
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "rtc_base/numerics/safe_conversions.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int ceil(int n, int m) {
+  return (n + m - 1) / m;
+}
+
+// Number of 10 ms frames required to fill a pitch buffer having size
+// `kBufSize24kHz`.
+constexpr int kNumTestDataFrames = ceil(kBufSize24kHz, kFrameSize10ms24kHz);
+// Number of samples for the test data.
+constexpr int kNumTestDataSize = kNumTestDataFrames * kFrameSize10ms24kHz;
+
+// Verifies that the pitch in Hz is in the detectable range.
+bool PitchIsValid(float pitch_hz) {
+  const int pitch_period = static_cast<float>(kSampleRate24kHz) / pitch_hz;
+  return kInitialMinPitch24kHz <= pitch_period &&
+         pitch_period <= kMaxPitch24kHz;
+}
+
+void CreatePureTone(float amplitude, float freq_hz, rtc::ArrayView<float> dst) {
+  for (int i = 0; rtc::SafeLt(i, dst.size()); ++i) {
+    dst[i] = amplitude * std::sin(2.f * kPi * freq_hz * i / kSampleRate24kHz);
+  }
+}
+
+// Feeds `features_extractor` with `samples` splitting it in 10 ms frames.
+// For every frame, the output is written into `feature_vector`. Returns true
+// if silence is detected in the last frame.
+bool FeedTestData(FeaturesExtractor& features_extractor,
+                  rtc::ArrayView<const float> samples,
+                  rtc::ArrayView<float, kFeatureVectorSize> feature_vector) {
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  bool is_silence = true;
+  const int num_frames = samples.size() / kFrameSize10ms24kHz;
+  for (int i = 0; i < num_frames; ++i) {
+    is_silence = features_extractor.CheckSilenceComputeFeatures(
+        {samples.data() + i * kFrameSize10ms24kHz, kFrameSize10ms24kHz},
+        feature_vector);
+  }
+  return is_silence;
+}
+
+// Extracts the features for two pure tones and verifies that the pitch field
+// values reflect the known tone frequencies.
+TEST(RnnVadTest, FeatureExtractionLowHighPitch) {
+  constexpr float amplitude = 1000.f;
+  constexpr float low_pitch_hz = 150.f;
+  constexpr float high_pitch_hz = 250.f;
+  ASSERT_TRUE(PitchIsValid(low_pitch_hz));
+  ASSERT_TRUE(PitchIsValid(high_pitch_hz));
+
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+  FeaturesExtractor features_extractor(cpu_features);
+  std::vector<float> samples(kNumTestDataSize);
+  std::vector<float> feature_vector(kFeatureVectorSize);
+  ASSERT_EQ(kFeatureVectorSize, rtc::dchecked_cast<int>(feature_vector.size()));
+  rtc::ArrayView<float, kFeatureVectorSize> feature_vector_view(
+      feature_vector.data(), kFeatureVectorSize);
+
+  // Extract the normalized scalar feature that is proportional to the estimated
+  // pitch period.
+  constexpr int pitch_feature_index = kFeatureVectorSize - 2;
+  // Low frequency tone - i.e., high period.
+  CreatePureTone(amplitude, low_pitch_hz, samples);
+  ASSERT_FALSE(FeedTestData(features_extractor, samples, feature_vector_view));
+  float high_pitch_period = feature_vector_view[pitch_feature_index];
+  // High frequency tone - i.e., low period.
+  features_extractor.Reset();
+  CreatePureTone(amplitude, high_pitch_hz, samples);
+  ASSERT_FALSE(FeedTestData(features_extractor, samples, feature_vector_view));
+  float low_pitch_period = feature_vector_view[pitch_feature_index];
+  // Check.
+  EXPECT_LT(low_pitch_period, high_pitch_period);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
new file mode 100644
index 0000000000..484bfba459
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
@@ -0,0 +1,141 @@
+/*
+ *  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/rnn_vad/lp_residual.h"
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Computes auto-correlation coefficients for `x` and writes them in
+// `auto_corr`. The lag values are in {0, ..., max_lag - 1}, where max_lag
+// equals the size of `auto_corr`.
+void ComputeAutoCorrelation(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kNumLpcCoefficients> auto_corr) {
+  constexpr int max_lag = auto_corr.size();
+  RTC_DCHECK_LT(max_lag, x.size());
+  for (int lag = 0; lag < max_lag; ++lag) {
+    auto_corr[lag] =
+        std::inner_product(x.begin(), x.end() - lag, x.begin() + lag, 0.f);
+  }
+}
+
+// Applies denoising to the auto-correlation coefficients.
+void DenoiseAutoCorrelation(
+    rtc::ArrayView<float, kNumLpcCoefficients> auto_corr) {
+  // Assume -40 dB white noise floor.
+  auto_corr[0] *= 1.0001f;
+  // Hard-coded values obtained as
+  // [np.float32((0.008*0.008*i*i)) for i in range(1,5)].
+  auto_corr[1] -= auto_corr[1] * 0.000064f;
+  auto_corr[2] -= auto_corr[2] * 0.000256f;
+  auto_corr[3] -= auto_corr[3] * 0.000576f;
+  auto_corr[4] -= auto_corr[4] * 0.001024f;
+  static_assert(kNumLpcCoefficients == 5, "Update `auto_corr`.");
+}
+
+// Computes the initial inverse filter coefficients given the auto-correlation
+// coefficients of an input frame.
+void ComputeInitialInverseFilterCoefficients(
+    rtc::ArrayView<const float, kNumLpcCoefficients> auto_corr,
+    rtc::ArrayView<float, kNumLpcCoefficients - 1> lpc_coeffs) {
+  float error = auto_corr[0];
+  for (int i = 0; i < kNumLpcCoefficients - 1; ++i) {
+    float reflection_coeff = 0.f;
+    for (int j = 0; j < i; ++j) {
+      reflection_coeff += lpc_coeffs[j] * auto_corr[i - j];
+    }
+    reflection_coeff += auto_corr[i + 1];
+
+    // Avoid division by numbers close to zero.
+    constexpr float kMinErrorMagnitude = 1e-6f;
+    if (std::fabs(error) < kMinErrorMagnitude) {
+      error = std::copysign(kMinErrorMagnitude, error);
+    }
+
+    reflection_coeff /= -error;
+    // Update LPC coefficients and total error.
+    lpc_coeffs[i] = reflection_coeff;
+    for (int j = 0; j < ((i + 1) >> 1); ++j) {
+      const float tmp1 = lpc_coeffs[j];
+      const float tmp2 = lpc_coeffs[i - 1 - j];
+      lpc_coeffs[j] = tmp1 + reflection_coeff * tmp2;
+      lpc_coeffs[i - 1 - j] = tmp2 + reflection_coeff * tmp1;
+    }
+    error -= reflection_coeff * reflection_coeff * error;
+    if (error < 0.001f * auto_corr[0]) {
+      break;
+    }
+  }
+}
+
+}  // namespace
+
+void ComputeAndPostProcessLpcCoefficients(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs) {
+  std::array<float, kNumLpcCoefficients> auto_corr;
+  ComputeAutoCorrelation(x, auto_corr);
+  if (auto_corr[0] == 0.f) {  // Empty frame.
+    std::fill(lpc_coeffs.begin(), lpc_coeffs.end(), 0);
+    return;
+  }
+  DenoiseAutoCorrelation(auto_corr);
+  std::array<float, kNumLpcCoefficients - 1> lpc_coeffs_pre{};
+  ComputeInitialInverseFilterCoefficients(auto_corr, lpc_coeffs_pre);
+  // LPC coefficients post-processing.
+  // TODO(bugs.webrtc.org/9076): Consider removing these steps.
+  lpc_coeffs_pre[0] *= 0.9f;
+  lpc_coeffs_pre[1] *= 0.9f * 0.9f;
+  lpc_coeffs_pre[2] *= 0.9f * 0.9f * 0.9f;
+  lpc_coeffs_pre[3] *= 0.9f * 0.9f * 0.9f * 0.9f;
+  constexpr float kC = 0.8f;
+  lpc_coeffs[0] = lpc_coeffs_pre[0] + kC;
+  lpc_coeffs[1] = lpc_coeffs_pre[1] + kC * lpc_coeffs_pre[0];
+  lpc_coeffs[2] = lpc_coeffs_pre[2] + kC * lpc_coeffs_pre[1];
+  lpc_coeffs[3] = lpc_coeffs_pre[3] + kC * lpc_coeffs_pre[2];
+  lpc_coeffs[4] = kC * lpc_coeffs_pre[3];
+  static_assert(kNumLpcCoefficients == 5, "Update `lpc_coeffs(_pre)`.");
+}
+
+void ComputeLpResidual(
+    rtc::ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float> y) {
+  RTC_DCHECK_GT(x.size(), kNumLpcCoefficients);
+  RTC_DCHECK_EQ(x.size(), y.size());
+  // The code below implements the following operation:
+  // y[i] = x[i] + dot_product({x[i], ..., x[i - kNumLpcCoefficients + 1]},
+  //                           lpc_coeffs)
+  // Edge case: i < kNumLpcCoefficients.
+  y[0] = x[0];
+  for (int i = 1; i < kNumLpcCoefficients; ++i) {
+    y[i] =
+        std::inner_product(x.crend() - i, x.crend(), lpc_coeffs.cbegin(), x[i]);
+  }
+  // Regular case.
+  auto last = x.crend();
+  for (int i = kNumLpcCoefficients; rtc::SafeLt(i, y.size()); ++i, --last) {
+    y[i] = std::inner_product(last - kNumLpcCoefficients, last,
+                              lpc_coeffs.cbegin(), x[i]);
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
new file mode 100644
index 0000000000..d04c536ec1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
@@ -0,0 +1,41 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
+
+#include <stddef.h>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Linear predictive coding (LPC) inverse filter length.
+constexpr int kNumLpcCoefficients = 5;
+
+// Given a frame `x`, computes a post-processed version of LPC coefficients
+// tailored for pitch estimation.
+void ComputeAndPostProcessLpcCoefficients(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs);
+
+// Computes the LP residual for the input frame `x` and the LPC coefficients
+// `lpc_coeffs`. `y` and `x` can point to the same array for in-place
+// computation.
+void ComputeLpResidual(
+    rtc::ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float> y);
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc
new file mode 100644
index 0000000000..7b3a4a3f65
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc
@@ -0,0 +1,80 @@
+/*
+ *  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/rnn_vad/lp_residual.h"
+
+#include <algorithm>
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Checks that the LP residual can be computed on an empty frame.
+TEST(RnnVadTest, LpResidualOfEmptyFrame) {
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+
+  // Input frame (empty, i.e., all samples set to 0).
+  std::array<float, kFrameSize10ms24kHz> empty_frame;
+  empty_frame.fill(0.f);
+  // Compute inverse filter coefficients.
+  std::array<float, kNumLpcCoefficients> lpc;
+  ComputeAndPostProcessLpcCoefficients(empty_frame, lpc);
+  // Compute LP residual.
+  std::array<float, kFrameSize10ms24kHz> lp_residual;
+  ComputeLpResidual(lpc, empty_frame, lp_residual);
+}
+
+// Checks that the computed LP residual is bit-exact given test input data.
+TEST(RnnVadTest, LpResidualPipelineBitExactness) {
+  // Input and expected output readers.
+  ChunksFileReader pitch_buffer_reader = CreatePitchBuffer24kHzReader();
+  ChunksFileReader lp_pitch_reader = CreateLpResidualAndPitchInfoReader();
+
+  // Buffers.
+  std::vector<float> pitch_buffer_24kHz(kBufSize24kHz);
+  std::array<float, kNumLpcCoefficients> lpc;
+  std::vector<float> computed_lp_residual(kBufSize24kHz);
+  std::vector<float> expected_lp_residual(kBufSize24kHz);
+
+  // Test length.
+  const int num_frames =
+      std::min(pitch_buffer_reader.num_chunks, 300);  // Max 3 s.
+  ASSERT_GE(lp_pitch_reader.num_chunks, num_frames);
+
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  for (int i = 0; i < num_frames; ++i) {
+    SCOPED_TRACE(i);
+    // Read input.
+    ASSERT_TRUE(pitch_buffer_reader.reader->ReadChunk(pitch_buffer_24kHz));
+    // Read expected output (ignore pitch gain and period).
+    ASSERT_TRUE(lp_pitch_reader.reader->ReadChunk(expected_lp_residual));
+    lp_pitch_reader.reader->SeekForward(2);  // Pitch period and strength.
+    // Check every 200 ms.
+    if (i % 20 == 0) {
+      ComputeAndPostProcessLpcCoefficients(pitch_buffer_24kHz, lpc);
+      ComputeLpResidual(lpc, pitch_buffer_24kHz, computed_lp_residual);
+      ExpectNearAbsolute(expected_lp_residual, computed_lp_residual, kFloatMin);
+    }
+  }
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
new file mode 100644
index 0000000000..419620fc0c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
@@ -0,0 +1,70 @@
+/*
+ *  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/rnn_vad/pitch_search.h"
+
+#include <array>
+#include <cstddef>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+PitchEstimator::PitchEstimator(const AvailableCpuFeatures& cpu_features)
+    : cpu_features_(cpu_features),
+      y_energy_24kHz_(kRefineNumLags24kHz, 0.f),
+      pitch_buffer_12kHz_(kBufSize12kHz),
+      auto_correlation_12kHz_(kNumLags12kHz) {}
+
+PitchEstimator::~PitchEstimator() = default;
+
+int PitchEstimator::Estimate(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer) {
+  rtc::ArrayView<float, kBufSize12kHz> pitch_buffer_12kHz_view(
+      pitch_buffer_12kHz_.data(), kBufSize12kHz);
+  RTC_DCHECK_EQ(pitch_buffer_12kHz_.size(), pitch_buffer_12kHz_view.size());
+  rtc::ArrayView<float, kNumLags12kHz> auto_correlation_12kHz_view(
+      auto_correlation_12kHz_.data(), kNumLags12kHz);
+  RTC_DCHECK_EQ(auto_correlation_12kHz_.size(),
+                auto_correlation_12kHz_view.size());
+
+  // TODO(bugs.chromium.org/10480): Use `cpu_features_` to estimate pitch.
+  // Perform the initial pitch search at 12 kHz.
+  Decimate2x(pitch_buffer, pitch_buffer_12kHz_view);
+  auto_corr_calculator_.ComputeOnPitchBuffer(pitch_buffer_12kHz_view,
+                                             auto_correlation_12kHz_view);
+  CandidatePitchPeriods pitch_periods = ComputePitchPeriod12kHz(
+      pitch_buffer_12kHz_view, auto_correlation_12kHz_view, cpu_features_);
+  // The refinement is done using the pitch buffer that contains 24 kHz samples.
+  // Therefore, adapt the inverted lags in `pitch_candidates_inv_lags` from 12
+  // to 24 kHz.
+  pitch_periods.best *= 2;
+  pitch_periods.second_best *= 2;
+
+  // Refine the initial pitch period estimation from 12 kHz to 48 kHz.
+  // Pre-compute frame energies at 24 kHz.
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_24kHz_view(
+      y_energy_24kHz_.data(), kRefineNumLags24kHz);
+  RTC_DCHECK_EQ(y_energy_24kHz_.size(), y_energy_24kHz_view.size());
+  ComputeSlidingFrameSquareEnergies24kHz(pitch_buffer, y_energy_24kHz_view,
+                                         cpu_features_);
+  // Estimation at 48 kHz.
+  const int pitch_lag_48kHz = ComputePitchPeriod48kHz(
+      pitch_buffer, y_energy_24kHz_view, pitch_periods, cpu_features_);
+  last_pitch_48kHz_ = ComputeExtendedPitchPeriod48kHz(
+      pitch_buffer, y_energy_24kHz_view,
+      /*initial_pitch_period_48kHz=*/kMaxPitch48kHz - pitch_lag_48kHz,
+      last_pitch_48kHz_, cpu_features_);
+  return last_pitch_48kHz_.period;
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h
new file mode 100644
index 0000000000..42c448eb56
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h
@@ -0,0 +1,54 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/auto_correlation.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+#include "rtc_base/gtest_prod_util.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Pitch estimator.
+class PitchEstimator {
+ public:
+  explicit PitchEstimator(const AvailableCpuFeatures& cpu_features);
+  PitchEstimator(const PitchEstimator&) = delete;
+  PitchEstimator& operator=(const PitchEstimator&) = delete;
+  ~PitchEstimator();
+  // Returns the estimated pitch period at 48 kHz.
+  int Estimate(rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer);
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(RnnVadTest, PitchSearchWithinTolerance);
+  float GetLastPitchStrengthForTesting() const {
+    return last_pitch_48kHz_.strength;
+  }
+
+  const AvailableCpuFeatures cpu_features_;
+  PitchInfo last_pitch_48kHz_{};
+  AutoCorrelationCalculator auto_corr_calculator_;
+  std::vector<float> y_energy_24kHz_;
+  std::vector<float> pitch_buffer_12kHz_;
+  std::vector<float> auto_correlation_12kHz_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
new file mode 100644
index 0000000000..e8c912518d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -0,0 +1,513 @@
+/*
+ *  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/rnn_vad/pitch_search_internal.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+#include <numeric>
+
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+float ComputeAutoCorrelation(
+    int inverted_lag,
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    const VectorMath& vector_math) {
+  RTC_DCHECK_LT(inverted_lag, kBufSize24kHz);
+  RTC_DCHECK_LT(inverted_lag, kRefineNumLags24kHz);
+  static_assert(kMaxPitch24kHz < kBufSize24kHz, "");
+  return vector_math.DotProduct(
+      pitch_buffer.subview(/*offset=*/kMaxPitch24kHz),
+      pitch_buffer.subview(inverted_lag, kFrameSize20ms24kHz));
+}
+
+// Given an auto-correlation coefficient `curr_auto_correlation` and its
+// neighboring values `prev_auto_correlation` and `next_auto_correlation`
+// computes a pseudo-interpolation offset to be applied to the pitch period
+// associated to `curr`. The output is a lag in {-1, 0, +1}.
+// TODO(bugs.webrtc.org/9076): Consider removing this method.
+// `GetPitchPseudoInterpolationOffset()` it is relevant only if the spectral
+// analysis works at a sample rate that is twice as that of the pitch buffer;
+// In particular, it is not relevant for the estimated pitch period feature fed
+// into the RNN.
+int GetPitchPseudoInterpolationOffset(float prev_auto_correlation,
+                                      float curr_auto_correlation,
+                                      float next_auto_correlation) {
+  if ((next_auto_correlation - prev_auto_correlation) >
+      0.7f * (curr_auto_correlation - prev_auto_correlation)) {
+    return 1;  // `next_auto_correlation` is the largest auto-correlation
+               // coefficient.
+  } else if ((prev_auto_correlation - next_auto_correlation) >
+             0.7f * (curr_auto_correlation - next_auto_correlation)) {
+    return -1;  // `prev_auto_correlation` is the largest auto-correlation
+                // coefficient.
+  }
+  return 0;
+}
+
+// Refines a pitch period `lag` encoded as lag with pseudo-interpolation. The
+// output sample rate is twice as that of `lag`.
+int PitchPseudoInterpolationLagPitchBuf(
+    int lag,
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    const VectorMath& vector_math) {
+  int offset = 0;
+  // Cannot apply pseudo-interpolation at the boundaries.
+  if (lag > 0 && lag < kMaxPitch24kHz) {
+    const int inverted_lag = kMaxPitch24kHz - lag;
+    offset = GetPitchPseudoInterpolationOffset(
+        ComputeAutoCorrelation(inverted_lag + 1, pitch_buffer, vector_math),
+        ComputeAutoCorrelation(inverted_lag, pitch_buffer, vector_math),
+        ComputeAutoCorrelation(inverted_lag - 1, pitch_buffer, vector_math));
+  }
+  return 2 * lag + offset;
+}
+
+// Integer multipliers used in ComputeExtendedPitchPeriod48kHz() when
+// looking for sub-harmonics.
+// The values have been chosen to serve the following algorithm. Given the
+// initial pitch period T, we examine whether one of its harmonics is the true
+// fundamental frequency. We consider T/k with k in {2, ..., 15}. For each of
+// these harmonics, in addition to the pitch strength of itself, we choose one
+// multiple of its pitch period, n*T/k, to validate it (by averaging their pitch
+// strengths). The multiplier n is chosen so that n*T/k is used only one time
+// over all k. When for example k = 4, we should also expect a peak at 3*T/4.
+// When k = 8 instead we don't want to look at 2*T/8, since we have already
+// checked T/4 before. Instead, we look at T*3/8.
+// The array can be generate in Python as follows:
+//   from fractions import Fraction
+//   # Smallest positive integer not in X.
+//   def mex(X):
+//     for i in range(1, int(max(X)+2)):
+//       if i not in X:
+//         return i
+//   # Visited multiples of the period.
+//   S = {1}
+//   for n in range(2, 16):
+//     sn = mex({n * i for i in S} | {1})
+//     S = S | {Fraction(1, n), Fraction(sn, n)}
+//     print(sn, end=', ')
+constexpr std::array<int, 14> kSubHarmonicMultipliers = {
+    {3, 2, 3, 2, 5, 2, 3, 2, 3, 2, 5, 2, 3, 2}};
+
+struct Range {
+  int min;
+  int max;
+};
+
+// Number of analyzed pitches to the left(right) of a pitch candidate.
+constexpr int kPitchNeighborhoodRadius = 2;
+
+// Creates a pitch period interval centered in `inverted_lag` with hard-coded
+// radius. Clipping is applied so that the interval is always valid for a 24 kHz
+// pitch buffer.
+Range CreateInvertedLagRange(int inverted_lag) {
+  return {std::max(inverted_lag - kPitchNeighborhoodRadius, 0),
+          std::min(inverted_lag + kPitchNeighborhoodRadius,
+                   kInitialNumLags24kHz - 1)};
+}
+
+constexpr int kNumPitchCandidates = 2;  // Best and second best.
+// Maximum number of analyzed pitch periods.
+constexpr int kMaxPitchPeriods24kHz =
+    kNumPitchCandidates * (2 * kPitchNeighborhoodRadius + 1);
+
+// Collection of inverted lags.
+class InvertedLagsIndex {
+ public:
+  InvertedLagsIndex() : num_entries_(0) {}
+  // Adds an inverted lag to the index. Cannot add more than
+  // `kMaxPitchPeriods24kHz` values.
+  void Append(int inverted_lag) {
+    RTC_DCHECK_LT(num_entries_, kMaxPitchPeriods24kHz);
+    inverted_lags_[num_entries_++] = inverted_lag;
+  }
+  const int* data() const { return inverted_lags_.data(); }
+  int size() const { return num_entries_; }
+
+ private:
+  std::array<int, kMaxPitchPeriods24kHz> inverted_lags_;
+  int num_entries_;
+};
+
+// Computes the auto correlation coefficients for the inverted lags in the
+// closed interval `inverted_lags`. Updates `inverted_lags_index` by appending
+// the inverted lags for the computed auto correlation values.
+void ComputeAutoCorrelation(
+    Range inverted_lags,
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<float, kInitialNumLags24kHz> auto_correlation,
+    InvertedLagsIndex& inverted_lags_index,
+    const VectorMath& vector_math) {
+  // Check valid range.
+  RTC_DCHECK_LE(inverted_lags.min, inverted_lags.max);
+  // Trick to avoid zero initialization of `auto_correlation`.
+  // Needed by the pseudo-interpolation.
+  if (inverted_lags.min > 0) {
+    auto_correlation[inverted_lags.min - 1] = 0.f;
+  }
+  if (inverted_lags.max < kInitialNumLags24kHz - 1) {
+    auto_correlation[inverted_lags.max + 1] = 0.f;
+  }
+  // Check valid `inverted_lag` indexes.
+  RTC_DCHECK_GE(inverted_lags.min, 0);
+  RTC_DCHECK_LT(inverted_lags.max, kInitialNumLags24kHz);
+  for (int inverted_lag = inverted_lags.min; inverted_lag <= inverted_lags.max;
+       ++inverted_lag) {
+    auto_correlation[inverted_lag] =
+        ComputeAutoCorrelation(inverted_lag, pitch_buffer, vector_math);
+    inverted_lags_index.Append(inverted_lag);
+  }
+}
+
+// Searches the strongest pitch period at 24 kHz and returns its inverted lag at
+// 48 kHz.
+int ComputePitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const int> inverted_lags,
+    rtc::ArrayView<const float, kInitialNumLags24kHz> auto_correlation,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    const VectorMath& vector_math) {
+  static_assert(kMaxPitch24kHz > kInitialNumLags24kHz, "");
+  static_assert(kMaxPitch24kHz < kBufSize24kHz, "");
+  int best_inverted_lag = 0;     // Pitch period.
+  float best_numerator = -1.f;   // Pitch strength numerator.
+  float best_denominator = 0.f;  // Pitch strength denominator.
+  for (int inverted_lag : inverted_lags) {
+    // A pitch candidate must have positive correlation.
+    if (auto_correlation[inverted_lag] > 0.f) {
+      // Auto-correlation energy normalized by frame energy.
+      const float numerator =
+          auto_correlation[inverted_lag] * auto_correlation[inverted_lag];
+      const float denominator = y_energy[inverted_lag];
+      // Compare numerator/denominator ratios without using divisions.
+      if (numerator * best_denominator > best_numerator * denominator) {
+        best_inverted_lag = inverted_lag;
+        best_numerator = numerator;
+        best_denominator = denominator;
+      }
+    }
+  }
+  // Pseudo-interpolation to transform `best_inverted_lag` (24 kHz pitch) to a
+  // 48 kHz pitch period.
+  if (best_inverted_lag == 0 || best_inverted_lag >= kInitialNumLags24kHz - 1) {
+    // Cannot apply pseudo-interpolation at the boundaries.
+    return best_inverted_lag * 2;
+  }
+  int offset = GetPitchPseudoInterpolationOffset(
+      auto_correlation[best_inverted_lag + 1],
+      auto_correlation[best_inverted_lag],
+      auto_correlation[best_inverted_lag - 1]);
+  // TODO(bugs.webrtc.org/9076): When retraining, check if `offset` below should
+  // be subtracted since `inverted_lag` is an inverted lag but offset is a lag.
+  return 2 * best_inverted_lag + offset;
+}
+
+// Returns an alternative pitch period for `pitch_period` given a `multiplier`
+// and a `divisor` of the period.
+constexpr int GetAlternativePitchPeriod(int pitch_period,
+                                        int multiplier,
+                                        int divisor) {
+  RTC_DCHECK_GT(divisor, 0);
+  // Same as `round(multiplier * pitch_period / divisor)`.
+  return (2 * multiplier * pitch_period + divisor) / (2 * divisor);
+}
+
+// Returns true if the alternative pitch period is stronger than the initial one
+// given the last estimated pitch and the value of `period_divisor` used to
+// compute the alternative pitch period via `GetAlternativePitchPeriod()`.
+bool IsAlternativePitchStrongerThanInitial(PitchInfo last,
+                                           PitchInfo initial,
+                                           PitchInfo alternative,
+                                           int period_divisor) {
+  // Initial pitch period candidate thresholds for a sample rate of 24 kHz.
+  // Computed as [5*k*k for k in range(16)].
+  constexpr std::array<int, 14> kInitialPitchPeriodThresholds = {
+      {20, 45, 80, 125, 180, 245, 320, 405, 500, 605, 720, 845, 980, 1125}};
+  static_assert(
+      kInitialPitchPeriodThresholds.size() == kSubHarmonicMultipliers.size(),
+      "");
+  RTC_DCHECK_GE(last.period, 0);
+  RTC_DCHECK_GE(initial.period, 0);
+  RTC_DCHECK_GE(alternative.period, 0);
+  RTC_DCHECK_GE(period_divisor, 2);
+  // Compute a term that lowers the threshold when `alternative.period` is close
+  // to the last estimated period `last.period` - i.e., pitch tracking.
+  float lower_threshold_term = 0.f;
+  if (std::abs(alternative.period - last.period) <= 1) {
+    // The candidate pitch period is within 1 sample from the last one.
+    // Make the candidate at `alternative.period` very easy to be accepted.
+    lower_threshold_term = last.strength;
+  } else if (std::abs(alternative.period - last.period) == 2 &&
+             initial.period >
+                 kInitialPitchPeriodThresholds[period_divisor - 2]) {
+    // The candidate pitch period is 2 samples far from the last one and the
+    // period `initial.period` (from which `alternative.period` has been
+    // derived) is greater than a threshold. Make `alternative.period` easy to
+    // be accepted.
+    lower_threshold_term = 0.5f * last.strength;
+  }
+  // Set the threshold based on the strength of the initial estimate
+  // `initial.period`. Also reduce the chance of false positives caused by a
+  // bias towards high frequencies (originating from short-term correlations).
+  float threshold =
+      std::max(0.3f, 0.7f * initial.strength - lower_threshold_term);
+  if (alternative.period < 3 * kMinPitch24kHz) {
+    // High frequency.
+    threshold = std::max(0.4f, 0.85f * initial.strength - lower_threshold_term);
+  } else if (alternative.period < 2 * kMinPitch24kHz) {
+    // Even higher frequency.
+    threshold = std::max(0.5f, 0.9f * initial.strength - lower_threshold_term);
+  }
+  return alternative.strength > threshold;
+}
+
+}  // namespace
+
+void Decimate2x(rtc::ArrayView<const float, kBufSize24kHz> src,
+                rtc::ArrayView<float, kBufSize12kHz> dst) {
+  // TODO(bugs.webrtc.org/9076): Consider adding anti-aliasing filter.
+  static_assert(2 * kBufSize12kHz == kBufSize24kHz, "");
+  for (int i = 0; i < kBufSize12kHz; ++i) {
+    dst[i] = src[2 * i];
+  }
+}
+
+void ComputeSlidingFrameSquareEnergies24kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<float, kRefineNumLags24kHz> y_energy,
+    AvailableCpuFeatures cpu_features) {
+  VectorMath vector_math(cpu_features);
+  static_assert(kFrameSize20ms24kHz < kBufSize24kHz, "");
+  const auto frame_20ms_view = pitch_buffer.subview(0, kFrameSize20ms24kHz);
+  float yy = vector_math.DotProduct(frame_20ms_view, frame_20ms_view);
+  y_energy[0] = yy;
+  static_assert(kMaxPitch24kHz - 1 + kFrameSize20ms24kHz < kBufSize24kHz, "");
+  static_assert(kMaxPitch24kHz < kRefineNumLags24kHz, "");
+  for (int inverted_lag = 0; inverted_lag < kMaxPitch24kHz; ++inverted_lag) {
+    yy -= pitch_buffer[inverted_lag] * pitch_buffer[inverted_lag];
+    yy += pitch_buffer[inverted_lag + kFrameSize20ms24kHz] *
+          pitch_buffer[inverted_lag + kFrameSize20ms24kHz];
+    yy = std::max(1.f, yy);
+    y_energy[inverted_lag + 1] = yy;
+  }
+}
+
+CandidatePitchPeriods ComputePitchPeriod12kHz(
+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buffer,
+    rtc::ArrayView<const float, kNumLags12kHz> auto_correlation,
+    AvailableCpuFeatures cpu_features) {
+  static_assert(kMaxPitch12kHz > kNumLags12kHz, "");
+  static_assert(kMaxPitch12kHz < kBufSize12kHz, "");
+
+  // Stores a pitch candidate period and strength information.
+  struct PitchCandidate {
+    // Pitch period encoded as inverted lag.
+    int period_inverted_lag = 0;
+    // Pitch strength encoded as a ratio.
+    float strength_numerator = -1.f;
+    float strength_denominator = 0.f;
+    // Compare the strength of two pitch candidates.
+    bool HasStrongerPitchThan(const PitchCandidate& b) const {
+      // Comparing the numerator/denominator ratios without using divisions.
+      return strength_numerator * b.strength_denominator >
+             b.strength_numerator * strength_denominator;
+    }
+  };
+
+  VectorMath vector_math(cpu_features);
+  static_assert(kFrameSize20ms12kHz + 1 < kBufSize12kHz, "");
+  const auto frame_view = pitch_buffer.subview(0, kFrameSize20ms12kHz + 1);
+  float denominator = 1.f + vector_math.DotProduct(frame_view, frame_view);
+  // Search best and second best pitches by looking at the scaled
+  // auto-correlation.
+  PitchCandidate best;
+  PitchCandidate second_best;
+  second_best.period_inverted_lag = 1;
+  for (int inverted_lag = 0; inverted_lag < kNumLags12kHz; ++inverted_lag) {
+    // A pitch candidate must have positive correlation.
+    if (auto_correlation[inverted_lag] > 0.f) {
+      PitchCandidate candidate{
+          inverted_lag,
+          auto_correlation[inverted_lag] * auto_correlation[inverted_lag],
+          denominator};
+      if (candidate.HasStrongerPitchThan(second_best)) {
+        if (candidate.HasStrongerPitchThan(best)) {
+          second_best = best;
+          best = candidate;
+        } else {
+          second_best = candidate;
+        }
+      }
+    }
+    // Update `squared_energy_y` for the next inverted lag.
+    const float y_old = pitch_buffer[inverted_lag];
+    const float y_new = pitch_buffer[inverted_lag + kFrameSize20ms12kHz];
+    denominator -= y_old * y_old;
+    denominator += y_new * y_new;
+    denominator = std::max(0.f, denominator);
+  }
+  return {best.period_inverted_lag, second_best.period_inverted_lag};
+}
+
+int ComputePitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    CandidatePitchPeriods pitch_candidates,
+    AvailableCpuFeatures cpu_features) {
+  // Compute the auto-correlation terms only for neighbors of the two pitch
+  // candidates (best and second best).
+  std::array<float, kInitialNumLags24kHz> auto_correlation;
+  InvertedLagsIndex inverted_lags_index;
+  // Create two inverted lag ranges so that `r1` precedes `r2`.
+  const bool swap_candidates =
+      pitch_candidates.best > pitch_candidates.second_best;
+  const Range r1 = CreateInvertedLagRange(
+      swap_candidates ? pitch_candidates.second_best : pitch_candidates.best);
+  const Range r2 = CreateInvertedLagRange(
+      swap_candidates ? pitch_candidates.best : pitch_candidates.second_best);
+  // Check valid ranges.
+  RTC_DCHECK_LE(r1.min, r1.max);
+  RTC_DCHECK_LE(r2.min, r2.max);
+  // Check `r1` precedes `r2`.
+  RTC_DCHECK_LE(r1.min, r2.min);
+  RTC_DCHECK_LE(r1.max, r2.max);
+  VectorMath vector_math(cpu_features);
+  if (r1.max + 1 >= r2.min) {
+    // Overlapping or adjacent ranges.
+    ComputeAutoCorrelation({r1.min, r2.max}, pitch_buffer, auto_correlation,
+                           inverted_lags_index, vector_math);
+  } else {
+    // Disjoint ranges.
+    ComputeAutoCorrelation(r1, pitch_buffer, auto_correlation,
+                           inverted_lags_index, vector_math);
+    ComputeAutoCorrelation(r2, pitch_buffer, auto_correlation,
+                           inverted_lags_index, vector_math);
+  }
+  return ComputePitchPeriod48kHz(pitch_buffer, inverted_lags_index,
+                                 auto_correlation, y_energy, vector_math);
+}
+
+PitchInfo ComputeExtendedPitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    int initial_pitch_period_48kHz,
+    PitchInfo last_pitch_48kHz,
+    AvailableCpuFeatures cpu_features) {
+  RTC_DCHECK_LE(kMinPitch48kHz, initial_pitch_period_48kHz);
+  RTC_DCHECK_LE(initial_pitch_period_48kHz, kMaxPitch48kHz);
+
+  // Stores information for a refined pitch candidate.
+  struct RefinedPitchCandidate {
+    int period;
+    float strength;
+    // Additional strength data used for the final pitch estimation.
+    float xy;        // Auto-correlation.
+    float y_energy;  // Energy of the sliding frame `y`.
+  };
+
+  const float x_energy = y_energy[kMaxPitch24kHz];
+  const auto pitch_strength = [x_energy](float xy, float y_energy) {
+    RTC_DCHECK_GE(x_energy * y_energy, 0.f);
+    return xy / std::sqrt(1.f + x_energy * y_energy);
+  };
+  VectorMath vector_math(cpu_features);
+
+  // Initialize the best pitch candidate with `initial_pitch_period_48kHz`.
+  RefinedPitchCandidate best_pitch;
+  best_pitch.period =
+      std::min(initial_pitch_period_48kHz / 2, kMaxPitch24kHz - 1);
+  best_pitch.xy = ComputeAutoCorrelation(kMaxPitch24kHz - best_pitch.period,
+                                         pitch_buffer, vector_math);
+  best_pitch.y_energy = y_energy[kMaxPitch24kHz - best_pitch.period];
+  best_pitch.strength = pitch_strength(best_pitch.xy, best_pitch.y_energy);
+  // Keep a copy of the initial pitch candidate.
+  const PitchInfo initial_pitch{best_pitch.period, best_pitch.strength};
+  // 24 kHz version of the last estimated pitch.
+  const PitchInfo last_pitch{last_pitch_48kHz.period / 2,
+                             last_pitch_48kHz.strength};
+
+  // Find `max_period_divisor` such that the result of
+  // `GetAlternativePitchPeriod(initial_pitch_period, 1, max_period_divisor)`
+  // equals `kMinPitch24kHz`.
+  const int max_period_divisor =
+      (2 * initial_pitch.period) / (2 * kMinPitch24kHz - 1);
+  for (int period_divisor = 2; period_divisor <= max_period_divisor;
+       ++period_divisor) {
+    PitchInfo alternative_pitch;
+    alternative_pitch.period = GetAlternativePitchPeriod(
+        initial_pitch.period, /*multiplier=*/1, period_divisor);
+    RTC_DCHECK_GE(alternative_pitch.period, kMinPitch24kHz);
+    // When looking at `alternative_pitch.period`, we also look at one of its
+    // sub-harmonics. `kSubHarmonicMultipliers` is used to know where to look.
+    // `period_divisor` == 2 is a special case since `dual_alternative_period`
+    // might be greater than the maximum pitch period.
+    int dual_alternative_period = GetAlternativePitchPeriod(
+        initial_pitch.period, kSubHarmonicMultipliers[period_divisor - 2],
+        period_divisor);
+    RTC_DCHECK_GT(dual_alternative_period, 0);
+    if (period_divisor == 2 && dual_alternative_period > kMaxPitch24kHz) {
+      dual_alternative_period = initial_pitch.period;
+    }
+    RTC_DCHECK_NE(alternative_pitch.period, dual_alternative_period)
+        << "The lower pitch period and the additional sub-harmonic must not "
+           "coincide.";
+    // Compute an auto-correlation score for the primary pitch candidate
+    // `alternative_pitch.period` by also looking at its possible sub-harmonic
+    // `dual_alternative_period`.
+    const float xy_primary_period = ComputeAutoCorrelation(
+        kMaxPitch24kHz - alternative_pitch.period, pitch_buffer, vector_math);
+    // TODO(webrtc:10480): Copy `xy_primary_period` if the secondary period is
+    // equal to the primary one.
+    const float xy_secondary_period = ComputeAutoCorrelation(
+        kMaxPitch24kHz - dual_alternative_period, pitch_buffer, vector_math);
+    const float xy = 0.5f * (xy_primary_period + xy_secondary_period);
+    const float yy =
+        0.5f * (y_energy[kMaxPitch24kHz - alternative_pitch.period] +
+                y_energy[kMaxPitch24kHz - dual_alternative_period]);
+    alternative_pitch.strength = pitch_strength(xy, yy);
+
+    // Maybe update best period.
+    if (IsAlternativePitchStrongerThanInitial(
+            last_pitch, initial_pitch, alternative_pitch, period_divisor)) {
+      best_pitch = {alternative_pitch.period, alternative_pitch.strength, xy,
+                    yy};
+    }
+  }
+
+  // Final pitch strength and period.
+  best_pitch.xy = std::max(0.f, best_pitch.xy);
+  RTC_DCHECK_LE(0.f, best_pitch.y_energy);
+  float final_pitch_strength =
+      (best_pitch.y_energy <= best_pitch.xy)
+          ? 1.f
+          : best_pitch.xy / (best_pitch.y_energy + 1.f);
+  final_pitch_strength = std::min(best_pitch.strength, final_pitch_strength);
+  int final_pitch_period_48kHz = std::max(
+      kMinPitch48kHz, PitchPseudoInterpolationLagPitchBuf(
+                          best_pitch.period, pitch_buffer, vector_math));
+
+  return {final_pitch_period_48kHz, final_pitch_strength};
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
new file mode 100644
index 0000000000..aa2dd13745
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
@@ -0,0 +1,114 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <utility>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Performs 2x decimation without any anti-aliasing filter.
+void Decimate2x(rtc::ArrayView<const float, kBufSize24kHz> src,
+                rtc::ArrayView<float, kBufSize12kHz> dst);
+
+// Key concepts and keywords used below in this file.
+//
+// The pitch estimation relies on a pitch buffer, which is an array-like data
+// structured designed as follows:
+//
+// |....A....|.....B.....|
+//
+// The part on the left, named `A` contains the oldest samples, whereas `B`
+// contains the most recent ones. The size of `A` corresponds to the maximum
+// pitch period, that of `B` to the analysis frame size (e.g., 16 ms and 20 ms
+// respectively).
+//
+// Pitch estimation is essentially based on the analysis of two 20 ms frames
+// extracted from the pitch buffer. One frame, called `x`, is kept fixed and
+// corresponds to `B` - i.e., the most recent 20 ms. The other frame, called
+// `y`, is extracted from different parts of the buffer instead.
+//
+// The offset between `x` and `y` corresponds to a specific pitch period.
+// For instance, if `y` is positioned at the beginning of the pitch buffer, then
+// the cross-correlation between `x` and `y` can be used as an indication of the
+// strength for the maximum pitch.
+//
+// Such an offset can be encoded in two ways:
+// - As a lag, which is the index in the pitch buffer for the first item in `y`
+// - As an inverted lag, which is the number of samples from the beginning of
+//   `x` and the end of `y`
+//
+// |---->| lag
+// |....A....|.....B.....|
+//       |<--| inverted lag
+//       |.....y.....| `y` 20 ms frame
+//
+// The inverted lag has the advantage of being directly proportional to the
+// corresponding pitch period.
+
+// Computes the sum of squared samples for every sliding frame `y` in the pitch
+// buffer. The indexes of `y_energy` are inverted lags.
+void ComputeSlidingFrameSquareEnergies24kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<float, kRefineNumLags24kHz> y_energy,
+    AvailableCpuFeatures cpu_features);
+
+// Top-2 pitch period candidates. Unit: number of samples - i.e., inverted lags.
+struct CandidatePitchPeriods {
+  int best;
+  int second_best;
+};
+
+// Computes the candidate pitch periods at 12 kHz given a view on the 12 kHz
+// pitch buffer and the auto-correlation values (having inverted lags as
+// indexes).
+CandidatePitchPeriods ComputePitchPeriod12kHz(
+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buffer,
+    rtc::ArrayView<const float, kNumLags12kHz> auto_correlation,
+    AvailableCpuFeatures cpu_features);
+
+// Computes the pitch period at 48 kHz given a view on the 24 kHz pitch buffer,
+// the energies for the sliding frames `y` at 24 kHz and the pitch period
+// candidates at 24 kHz (encoded as inverted lag).
+int ComputePitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    CandidatePitchPeriods pitch_candidates_24kHz,
+    AvailableCpuFeatures cpu_features);
+
+struct PitchInfo {
+  int period;
+  float strength;
+};
+
+// Computes the pitch period at 48 kHz searching in an extended pitch range
+// given a view on the 24 kHz pitch buffer, the energies for the sliding frames
+// `y` at 24 kHz, the initial 48 kHz estimation (computed by
+// `ComputePitchPeriod48kHz()`) and the last estimated pitch.
+PitchInfo ComputeExtendedPitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    int initial_pitch_period_48kHz,
+    PitchInfo last_pitch_48kHz,
+    AvailableCpuFeatures cpu_features);
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
new file mode 100644
index 0000000000..2a6e68f157
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
@@ -0,0 +1,217 @@
+/*
+ *  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/rnn_vad/pitch_search_internal.h"
+
+#include <array>
+#include <string>
+#include <tuple>
+
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "rtc_base/strings/string_builder.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kTestPitchPeriodsLow = 3 * kMinPitch48kHz / 2;
+constexpr int kTestPitchPeriodsHigh = (3 * kMinPitch48kHz + kMaxPitch48kHz) / 2;
+
+constexpr float kTestPitchStrengthLow = 0.35f;
+constexpr float kTestPitchStrengthHigh = 0.75f;
+
+template <class T>
+std::string PrintTestIndexAndCpuFeatures(
+    const ::testing::TestParamInfo<T>& info) {
+  rtc::StringBuilder builder;
+  builder << info.index << "_" << info.param.cpu_features.ToString();
+  return builder.str();
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  return v;
+}
+
+// Checks that the frame-wise sliding square energy function produces output
+// within tolerance given test input data.
+TEST(RnnVadTest, ComputeSlidingFrameSquareEnergies24kHzWithinTolerance) {
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+
+  PitchTestData test_data;
+  std::array<float, kRefineNumLags24kHz> computed_output;
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         computed_output, cpu_features);
+  auto square_energies_view = test_data.SquareEnergies24kHzView();
+  ExpectNearAbsolute({square_energies_view.data(), square_energies_view.size()},
+                     computed_output, 1e-3f);
+}
+
+// Checks that the estimated pitch period is bit-exact given test input data.
+TEST(RnnVadTest, ComputePitchPeriod12kHzBitExactness) {
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+
+  PitchTestData test_data;
+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
+  Decimate2x(test_data.PitchBuffer24kHzView(), pitch_buf_decimated);
+  CandidatePitchPeriods pitch_candidates;
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  pitch_candidates = ComputePitchPeriod12kHz(
+      pitch_buf_decimated, test_data.AutoCorrelation12kHzView(), cpu_features);
+  EXPECT_EQ(pitch_candidates.best, 140);
+  EXPECT_EQ(pitch_candidates.second_best, 142);
+}
+
+// Checks that the refined pitch period is bit-exact given test input data.
+TEST(RnnVadTest, ComputePitchPeriod48kHzBitExactness) {
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+
+  PitchTestData test_data;
+  std::vector<float> y_energy(kRefineNumLags24kHz);
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+                                                           kRefineNumLags24kHz);
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         y_energy_view, cpu_features);
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  EXPECT_EQ(
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              /*pitch_candidates=*/{280, 284}, cpu_features),
+      560);
+  EXPECT_EQ(
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              /*pitch_candidates=*/{260, 284}, cpu_features),
+      568);
+}
+
+struct PitchCandidatesParameters {
+  CandidatePitchPeriods pitch_candidates;
+  AvailableCpuFeatures cpu_features;
+};
+
+class PitchCandidatesParametrization
+    : public ::testing::TestWithParam<PitchCandidatesParameters> {};
+
+// Checks that the result of `ComputePitchPeriod48kHz()` does not depend on the
+// order of the input pitch candidates.
+TEST_P(PitchCandidatesParametrization,
+       ComputePitchPeriod48kHzOrderDoesNotMatter) {
+  const PitchCandidatesParameters params = GetParam();
+  const CandidatePitchPeriods swapped_pitch_candidates{
+      params.pitch_candidates.second_best, params.pitch_candidates.best};
+
+  PitchTestData test_data;
+  std::vector<float> y_energy(kRefineNumLags24kHz);
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+                                                           kRefineNumLags24kHz);
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         y_energy_view, params.cpu_features);
+  EXPECT_EQ(
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              params.pitch_candidates, params.cpu_features),
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              swapped_pitch_candidates, params.cpu_features));
+}
+
+std::vector<PitchCandidatesParameters> CreatePitchCandidatesParameters() {
+  std::vector<PitchCandidatesParameters> v;
+  for (AvailableCpuFeatures cpu_features : GetCpuFeaturesToTest()) {
+    v.push_back({{0, 2}, cpu_features});
+    v.push_back({{260, 284}, cpu_features});
+    v.push_back({{280, 284}, cpu_features});
+    v.push_back(
+        {{kInitialNumLags24kHz - 2, kInitialNumLags24kHz - 1}, cpu_features});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    PitchCandidatesParametrization,
+    ::testing::ValuesIn(CreatePitchCandidatesParameters()),
+    PrintTestIndexAndCpuFeatures<PitchCandidatesParameters>);
+
+struct ExtendedPitchPeriodSearchParameters {
+  int initial_pitch_period;
+  PitchInfo last_pitch;
+  PitchInfo expected_pitch;
+  AvailableCpuFeatures cpu_features;
+};
+
+class ExtendedPitchPeriodSearchParametrizaion
+    : public ::testing::TestWithParam<ExtendedPitchPeriodSearchParameters> {};
+
+// Checks that the computed pitch period is bit-exact and that the computed
+// pitch strength is within tolerance given test input data.
+TEST_P(ExtendedPitchPeriodSearchParametrizaion,
+       PeriodBitExactnessGainWithinTolerance) {
+  const ExtendedPitchPeriodSearchParameters params = GetParam();
+
+  PitchTestData test_data;
+  std::vector<float> y_energy(kRefineNumLags24kHz);
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+                                                           kRefineNumLags24kHz);
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         y_energy_view, params.cpu_features);
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  const auto computed_output = ComputeExtendedPitchPeriod48kHz(
+      test_data.PitchBuffer24kHzView(), y_energy_view,
+      params.initial_pitch_period, params.last_pitch, params.cpu_features);
+  EXPECT_EQ(params.expected_pitch.period, computed_output.period);
+  EXPECT_NEAR(params.expected_pitch.strength, computed_output.strength, 1e-6f);
+}
+
+std::vector<ExtendedPitchPeriodSearchParameters>
+CreateExtendedPitchPeriodSearchParameters() {
+  std::vector<ExtendedPitchPeriodSearchParameters> v;
+  for (AvailableCpuFeatures cpu_features : GetCpuFeaturesToTest()) {
+    for (int last_pitch_period :
+         {kTestPitchPeriodsLow, kTestPitchPeriodsHigh}) {
+      for (float last_pitch_strength :
+           {kTestPitchStrengthLow, kTestPitchStrengthHigh}) {
+        v.push_back({kTestPitchPeriodsLow,
+                     {last_pitch_period, last_pitch_strength},
+                     {91, -0.0188608f},
+                     cpu_features});
+        v.push_back({kTestPitchPeriodsHigh,
+                     {last_pitch_period, last_pitch_strength},
+                     {475, -0.0904344f},
+                     cpu_features});
+      }
+    }
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    ExtendedPitchPeriodSearchParametrizaion,
+    ::testing::ValuesIn(CreateExtendedPitchPeriodSearchParameters()),
+    PrintTestIndexAndCpuFeatures<ExtendedPitchPeriodSearchParameters>);
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
new file mode 100644
index 0000000000..79b44b995c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
@@ -0,0 +1,53 @@
+/*
+ *  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/rnn_vad/pitch_search.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Checks that the computed pitch period is bit-exact and that the computed
+// pitch gain is within tolerance given test input data.
+TEST(RnnVadTest, PitchSearchWithinTolerance) {
+  ChunksFileReader reader = CreateLpResidualAndPitchInfoReader();
+  const int num_frames = std::min(reader.num_chunks, 300);  // Max 3 s.
+  std::vector<float> lp_residual(kBufSize24kHz);
+  float expected_pitch_period, expected_pitch_strength;
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+  PitchEstimator pitch_estimator(cpu_features);
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    for (int i = 0; i < num_frames; ++i) {
+      SCOPED_TRACE(i);
+      ASSERT_TRUE(reader.reader->ReadChunk(lp_residual));
+      ASSERT_TRUE(reader.reader->ReadValue(expected_pitch_period));
+      ASSERT_TRUE(reader.reader->ReadValue(expected_pitch_strength));
+      int pitch_period =
+          pitch_estimator.Estimate({lp_residual.data(), kBufSize24kHz});
+      EXPECT_EQ(expected_pitch_period, pitch_period);
+      EXPECT_NEAR(expected_pitch_strength,
+                  pitch_estimator.GetLastPitchStrengthForTesting(), 15e-6f);
+    }
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
new file mode 100644
index 0000000000..a6f7fdd1a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
@@ -0,0 +1,65 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
+
+#include <array>
+#include <cstring>
+#include <type_traits>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Ring buffer for N arrays of type T each one with size S.
+template <typename T, int S, int N>
+class RingBuffer {
+  static_assert(S > 0, "");
+  static_assert(N > 0, "");
+  static_assert(std::is_arithmetic<T>::value,
+                "Integral or floating point required.");
+
+ public:
+  RingBuffer() : tail_(0) {}
+  RingBuffer(const RingBuffer&) = delete;
+  RingBuffer& operator=(const RingBuffer&) = delete;
+  ~RingBuffer() = default;
+  // Set the ring buffer values to zero.
+  void Reset() { buffer_.fill(0); }
+  // Replace the least recently pushed array in the buffer with `new_values`.
+  void Push(rtc::ArrayView<const T, S> new_values) {
+    std::memcpy(buffer_.data() + S * tail_, new_values.data(), S * sizeof(T));
+    tail_ += 1;
+    if (tail_ == N)
+      tail_ = 0;
+  }
+  // Return an array view onto the array with a given delay. A view on the last
+  // and least recently push array is returned when `delay` is 0 and N - 1
+  // respectively.
+  rtc::ArrayView<const T, S> GetArrayView(int delay) const {
+    RTC_DCHECK_LE(0, delay);
+    RTC_DCHECK_LT(delay, N);
+    int offset = tail_ - 1 - delay;
+    if (offset < 0)
+      offset += N;
+    return {buffer_.data() + S * offset, S};
+  }
+
+ private:
+  int tail_;  // Index of the least recently pushed sub-array.
+  std::array<T, S * N> buffer_{};
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
new file mode 100644
index 0000000000..d11d4eac3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
@@ -0,0 +1,112 @@
+/*
+ *  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/rnn_vad/ring_buffer.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Compare the elements of two given array views.
+template <typename T, std::ptrdiff_t S>
+void ExpectEq(rtc::ArrayView<const T, S> a, rtc::ArrayView<const T, S> b) {
+  for (int i = 0; i < S; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_EQ(a[i], b[i]);
+  }
+}
+
+// Test push/read sequences.
+template <typename T, int S, int N>
+void TestRingBuffer() {
+  SCOPED_TRACE(N);
+  SCOPED_TRACE(S);
+  std::array<T, S> prev_pushed_array;
+  std::array<T, S> pushed_array;
+  rtc::ArrayView<const T, S> pushed_array_view(pushed_array.data(), S);
+
+  // Init.
+  RingBuffer<T, S, N> ring_buf;
+  ring_buf.GetArrayView(0);
+  pushed_array.fill(0);
+  ring_buf.Push(pushed_array_view);
+  ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
+
+  // Push N times and check most recent and second most recent.
+  for (T v = 1; v <= static_cast<T>(N); ++v) {
+    SCOPED_TRACE(v);
+    prev_pushed_array = pushed_array;
+    pushed_array.fill(v);
+    ring_buf.Push(pushed_array_view);
+    ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
+    if (N > 1) {
+      pushed_array.fill(v - 1);
+      ExpectEq(pushed_array_view, ring_buf.GetArrayView(1));
+    }
+  }
+
+  // Check buffer.
+  for (int delay = 2; delay < N; ++delay) {
+    SCOPED_TRACE(delay);
+    T expected_value = N - static_cast<T>(delay);
+    pushed_array.fill(expected_value);
+    ExpectEq(pushed_array_view, ring_buf.GetArrayView(delay));
+  }
+}
+
+// Check that for different delays, different views are returned.
+TEST(RnnVadTest, RingBufferArrayViews) {
+  constexpr int s = 3;
+  constexpr int n = 4;
+  RingBuffer<int, s, n> ring_buf;
+  std::array<int, s> pushed_array;
+  pushed_array.fill(1);
+  for (int k = 0; k <= n; ++k) {  // Push data n + 1 times.
+    SCOPED_TRACE(k);
+    // Check array views.
+    for (int i = 0; i < n; ++i) {
+      SCOPED_TRACE(i);
+      auto view_i = ring_buf.GetArrayView(i);
+      for (int j = i + 1; j < n; ++j) {
+        SCOPED_TRACE(j);
+        auto view_j = ring_buf.GetArrayView(j);
+        EXPECT_NE(view_i, view_j);
+      }
+    }
+    ring_buf.Push(pushed_array);
+  }
+}
+
+TEST(RnnVadTest, RingBufferUnsigned) {
+  TestRingBuffer<uint8_t, 1, 1>();
+  TestRingBuffer<uint8_t, 2, 5>();
+  TestRingBuffer<uint8_t, 5, 2>();
+  TestRingBuffer<uint8_t, 5, 5>();
+}
+
+TEST(RnnVadTest, RingBufferSigned) {
+  TestRingBuffer<int, 1, 1>();
+  TestRingBuffer<int, 2, 5>();
+  TestRingBuffer<int, 5, 2>();
+  TestRingBuffer<int, 5, 5>();
+}
+
+TEST(RnnVadTest, RingBufferFloating) {
+  TestRingBuffer<float, 1, 1>();
+  TestRingBuffer<float, 2, 5>();
+  TestRingBuffer<float, 5, 2>();
+  TestRingBuffer<float, 5, 5>();
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
new file mode 100644
index 0000000000..475bef9775
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
@@ -0,0 +1,91 @@
+/*
+ *  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/rnn_vad/rnn.h"
+
+#include "rtc_base/checks.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+using ::rnnoise::kInputLayerInputSize;
+static_assert(kFeatureVectorSize == kInputLayerInputSize, "");
+using ::rnnoise::kInputDenseBias;
+using ::rnnoise::kInputDenseWeights;
+using ::rnnoise::kInputLayerOutputSize;
+static_assert(kInputLayerOutputSize <= kFullyConnectedLayerMaxUnits, "");
+
+using ::rnnoise::kHiddenGruBias;
+using ::rnnoise::kHiddenGruRecurrentWeights;
+using ::rnnoise::kHiddenGruWeights;
+using ::rnnoise::kHiddenLayerOutputSize;
+static_assert(kHiddenLayerOutputSize <= kGruLayerMaxUnits, "");
+
+using ::rnnoise::kOutputDenseBias;
+using ::rnnoise::kOutputDenseWeights;
+using ::rnnoise::kOutputLayerOutputSize;
+static_assert(kOutputLayerOutputSize <= kFullyConnectedLayerMaxUnits, "");
+
+}  // namespace
+
+RnnVad::RnnVad(const AvailableCpuFeatures& cpu_features)
+    : input_(kInputLayerInputSize,
+             kInputLayerOutputSize,
+             kInputDenseBias,
+             kInputDenseWeights,
+             ActivationFunction::kTansigApproximated,
+             cpu_features,
+             /*layer_name=*/"FC1"),
+      hidden_(kInputLayerOutputSize,
+              kHiddenLayerOutputSize,
+              kHiddenGruBias,
+              kHiddenGruWeights,
+              kHiddenGruRecurrentWeights,
+              cpu_features,
+              /*layer_name=*/"GRU1"),
+      output_(kHiddenLayerOutputSize,
+              kOutputLayerOutputSize,
+              kOutputDenseBias,
+              kOutputDenseWeights,
+              ActivationFunction::kSigmoidApproximated,
+              // The output layer is just 24x1. The unoptimized code is faster.
+              NoAvailableCpuFeatures(),
+              /*layer_name=*/"FC2") {
+  // Input-output chaining size checks.
+  RTC_DCHECK_EQ(input_.size(), hidden_.input_size())
+      << "The input and the hidden layers sizes do not match.";
+  RTC_DCHECK_EQ(hidden_.size(), output_.input_size())
+      << "The hidden and the output layers sizes do not match.";
+}
+
+RnnVad::~RnnVad() = default;
+
+void RnnVad::Reset() {
+  hidden_.Reset();
+}
+
+float RnnVad::ComputeVadProbability(
+    rtc::ArrayView<const float, kFeatureVectorSize> feature_vector,
+    bool is_silence) {
+  if (is_silence) {
+    Reset();
+    return 0.f;
+  }
+  input_.ComputeOutput(feature_vector);
+  hidden_.ComputeOutput(input_);
+  output_.ComputeOutput(hidden_);
+  RTC_DCHECK_EQ(output_.size(), 1);
+  return output_.data()[0];
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
new file mode 100644
index 0000000000..3148f1b3ff
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
@@ -0,0 +1,53 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
+
+#include <stddef.h>
+#include <sys/types.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn_gru.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Recurrent network with hard-coded architecture and weights for voice activity
+// detection.
+class RnnVad {
+ public:
+  explicit RnnVad(const AvailableCpuFeatures& cpu_features);
+  RnnVad(const RnnVad&) = delete;
+  RnnVad& operator=(const RnnVad&) = delete;
+  ~RnnVad();
+  void Reset();
+  // Observes `feature_vector` and `is_silence`, updates the RNN and returns the
+  // current voice probability.
+  float ComputeVadProbability(
+      rtc::ArrayView<const float, kFeatureVectorSize> feature_vector,
+      bool is_silence);
+
+ private:
+  FullyConnectedLayer input_;
+  GatedRecurrentLayer hidden_;
+  FullyConnectedLayer output_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
new file mode 100644
index 0000000000..a13e77461a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2020 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/rnn_vad/rnn_fc.h"
+
+#include <algorithm>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "third_party/rnnoise/src/rnn_activations.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+std::vector<float> GetScaledParams(rtc::ArrayView<const int8_t> params) {
+  std::vector<float> scaled_params(params.size());
+  std::transform(params.begin(), params.end(), scaled_params.begin(),
+                 [](int8_t x) -> float {
+                   return ::rnnoise::kWeightsScale * static_cast<float>(x);
+                 });
+  return scaled_params;
+}
+
+// TODO(bugs.chromium.org/10480): Hard-code optimized layout and remove this
+// function to improve setup time.
+// Casts and scales `weights` and re-arranges the layout.
+std::vector<float> PreprocessWeights(rtc::ArrayView<const int8_t> weights,
+                                     int output_size) {
+  if (output_size == 1) {
+    return GetScaledParams(weights);
+  }
+  // Transpose, scale and cast.
+  const int input_size = rtc::CheckedDivExact(
+      rtc::dchecked_cast<int>(weights.size()), output_size);
+  std::vector<float> w(weights.size());
+  for (int o = 0; o < output_size; ++o) {
+    for (int i = 0; i < input_size; ++i) {
+      w[o * input_size + i] = rnnoise::kWeightsScale *
+                              static_cast<float>(weights[i * output_size + o]);
+    }
+  }
+  return w;
+}
+
+rtc::FunctionView<float(float)> GetActivationFunction(
+    ActivationFunction activation_function) {
+  switch (activation_function) {
+    case ActivationFunction::kTansigApproximated:
+      return ::rnnoise::TansigApproximated;
+    case ActivationFunction::kSigmoidApproximated:
+      return ::rnnoise::SigmoidApproximated;
+  }
+}
+
+}  // namespace
+
+FullyConnectedLayer::FullyConnectedLayer(
+    const int input_size,
+    const int output_size,
+    const rtc::ArrayView<const int8_t> bias,
+    const rtc::ArrayView<const int8_t> weights,
+    ActivationFunction activation_function,
+    const AvailableCpuFeatures& cpu_features,
+    absl::string_view layer_name)
+    : input_size_(input_size),
+      output_size_(output_size),
+      bias_(GetScaledParams(bias)),
+      weights_(PreprocessWeights(weights, output_size)),
+      vector_math_(cpu_features),
+      activation_function_(GetActivationFunction(activation_function)) {
+  RTC_DCHECK_LE(output_size_, kFullyConnectedLayerMaxUnits)
+      << "Insufficient FC layer over-allocation (" << layer_name << ").";
+  RTC_DCHECK_EQ(output_size_, bias_.size())
+      << "Mismatching output size and bias terms array size (" << layer_name
+      << ").";
+  RTC_DCHECK_EQ(input_size_ * output_size_, weights_.size())
+      << "Mismatching input-output size and weight coefficients array size ("
+      << layer_name << ").";
+}
+
+FullyConnectedLayer::~FullyConnectedLayer() = default;
+
+void FullyConnectedLayer::ComputeOutput(rtc::ArrayView<const float> input) {
+  RTC_DCHECK_EQ(input.size(), input_size_);
+  rtc::ArrayView<const float> weights(weights_);
+  for (int o = 0; o < output_size_; ++o) {
+    output_[o] = activation_function_(
+        bias_[o] + vector_math_.DotProduct(
+                       input, weights.subview(o * input_size_, input_size_)));
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
new file mode 100644
index 0000000000..d23957a6f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_FC_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_FC_H_
+
+#include <array>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/function_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Activation function for a neural network cell.
+enum class ActivationFunction { kTansigApproximated, kSigmoidApproximated };
+
+// Maximum number of units for an FC layer.
+constexpr int kFullyConnectedLayerMaxUnits = 24;
+
+// Fully-connected layer with a custom activation function which owns the output
+// buffer.
+class FullyConnectedLayer {
+ public:
+  // Ctor. `output_size` cannot be greater than `kFullyConnectedLayerMaxUnits`.
+  FullyConnectedLayer(int input_size,
+                      int output_size,
+                      rtc::ArrayView<const int8_t> bias,
+                      rtc::ArrayView<const int8_t> weights,
+                      ActivationFunction activation_function,
+                      const AvailableCpuFeatures& cpu_features,
+                      absl::string_view layer_name);
+  FullyConnectedLayer(const FullyConnectedLayer&) = delete;
+  FullyConnectedLayer& operator=(const FullyConnectedLayer&) = delete;
+  ~FullyConnectedLayer();
+
+  // Returns the size of the input vector.
+  int input_size() const { return input_size_; }
+  // Returns the pointer to the first element of the output buffer.
+  const float* data() const { return output_.data(); }
+  // Returns the size of the output buffer.
+  int size() const { return output_size_; }
+
+  // Computes the fully-connected layer output.
+  void ComputeOutput(rtc::ArrayView<const float> input);
+
+ private:
+  const int input_size_;
+  const int output_size_;
+  const std::vector<float> bias_;
+  const std::vector<float> weights_;
+  const VectorMath vector_math_;
+  rtc::FunctionView<float(float)> activation_function_;
+  // Over-allocated array with size equal to `output_size_`.
+  std::array<float, kFullyConnectedLayerMaxUnits> output_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_FC_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
new file mode 100644
index 0000000000..ff9bb18bc2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2020 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/rnn_vad/rnn_fc.h"
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/test/performance_timer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/arch.h"
+#include "test/gtest.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+using ::rnnoise::kInputDenseBias;
+using ::rnnoise::kInputDenseWeights;
+using ::rnnoise::kInputLayerInputSize;
+using ::rnnoise::kInputLayerOutputSize;
+
+// Fully connected layer test data.
+constexpr std::array<float, 42> kFullyConnectedInputVector = {
+    -1.00131f,   -0.627069f, -7.81097f,  7.86285f,    -2.87145f,  3.32365f,
+    -0.653161f,  0.529839f,  -0.425307f, 0.25583f,    0.235094f,  0.230527f,
+    -0.144687f,  0.182785f,  0.57102f,   0.125039f,   0.479482f,  -0.0255439f,
+    -0.0073141f, -0.147346f, -0.217106f, -0.0846906f, -8.34943f,  3.09065f,
+    1.42628f,    -0.85235f,  -0.220207f, -0.811163f,  2.09032f,   -2.01425f,
+    -0.690268f,  -0.925327f, -0.541354f, 0.58455f,    -0.606726f, -0.0372358f,
+    0.565991f,   0.435854f,  0.420812f,  0.162198f,   -2.13f,     10.0089f};
+constexpr std::array<float, 24> kFullyConnectedExpectedOutput = {
+    -0.623293f, -0.988299f, 0.999378f,  0.967168f,  0.103087f,  -0.978545f,
+    -0.856347f, 0.346675f,  1.f,        -0.717442f, -0.544176f, 0.960363f,
+    0.983443f,  0.999991f,  -0.824335f, 0.984742f,  0.990208f,  0.938179f,
+    0.875092f,  0.999846f,  0.997707f,  -0.999382f, 0.973153f,  -0.966605f};
+
+class RnnFcParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+// Checks that the output of a fully connected layer is within tolerance given
+// test input data.
+TEST_P(RnnFcParametrization, CheckFullyConnectedLayerOutput) {
+  FullyConnectedLayer fc(kInputLayerInputSize, kInputLayerOutputSize,
+                         kInputDenseBias, kInputDenseWeights,
+                         ActivationFunction::kTansigApproximated,
+                         /*cpu_features=*/GetParam(),
+                         /*layer_name=*/"FC");
+  fc.ComputeOutput(kFullyConnectedInputVector);
+  ExpectNearAbsolute(kFullyConnectedExpectedOutput, fc, 1e-5f);
+}
+
+TEST_P(RnnFcParametrization, DISABLED_BenchmarkFullyConnectedLayer) {
+  const AvailableCpuFeatures cpu_features = GetParam();
+  FullyConnectedLayer fc(kInputLayerInputSize, kInputLayerOutputSize,
+                         kInputDenseBias, kInputDenseWeights,
+                         ActivationFunction::kTansigApproximated, cpu_features,
+                         /*layer_name=*/"FC");
+
+  constexpr int kNumTests = 10000;
+  ::webrtc::test::PerformanceTimer perf_timer(kNumTests);
+  for (int k = 0; k < kNumTests; ++k) {
+    perf_timer.StartTimer();
+    fc.ComputeOutput(kFullyConnectedInputVector);
+    perf_timer.StopTimer();
+  }
+  RTC_LOG(LS_INFO) << "CPU features: " << cpu_features.ToString() << " | "
+                   << (perf_timer.GetDurationAverage() / 1000) << " +/- "
+                   << (perf_timer.GetDurationStandardDeviation() / 1000)
+                   << " ms";
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    RnnFcParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
new file mode 100644
index 0000000000..ef37410caa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
@@ -0,0 +1,198 @@
+/*
+ *  Copyright (c) 2020 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/rnn_vad/rnn_gru.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "third_party/rnnoise/src/rnn_activations.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kNumGruGates = 3;  // Update, reset, output.
+
+std::vector<float> PreprocessGruTensor(rtc::ArrayView<const int8_t> tensor_src,
+                                       int output_size) {
+  // Transpose, cast and scale.
+  // `n` is the size of the first dimension of the 3-dim tensor `weights`.
+  const int n = rtc::CheckedDivExact(rtc::dchecked_cast<int>(tensor_src.size()),
+                                     output_size * kNumGruGates);
+  const int stride_src = kNumGruGates * output_size;
+  const int stride_dst = n * output_size;
+  std::vector<float> tensor_dst(tensor_src.size());
+  for (int g = 0; g < kNumGruGates; ++g) {
+    for (int o = 0; o < output_size; ++o) {
+      for (int i = 0; i < n; ++i) {
+        tensor_dst[g * stride_dst + o * n + i] =
+            ::rnnoise::kWeightsScale *
+            static_cast<float>(
+                tensor_src[i * stride_src + g * output_size + o]);
+      }
+    }
+  }
+  return tensor_dst;
+}
+
+// Computes the output for the update or the reset gate.
+// Operation: `g = sigmoid(W^T∙i + R^T∙s + b)` where
+// - `g`: output gate vector
+// - `W`: weights matrix
+// - `i`: input vector
+// - `R`: recurrent weights matrix
+// - `s`: state gate vector
+// - `b`: bias vector
+void ComputeUpdateResetGate(int input_size,
+                            int output_size,
+                            const VectorMath& vector_math,
+                            rtc::ArrayView<const float> input,
+                            rtc::ArrayView<const float> state,
+                            rtc::ArrayView<const float> bias,
+                            rtc::ArrayView<const float> weights,
+                            rtc::ArrayView<const float> recurrent_weights,
+                            rtc::ArrayView<float> gate) {
+  RTC_DCHECK_EQ(input.size(), input_size);
+  RTC_DCHECK_EQ(state.size(), output_size);
+  RTC_DCHECK_EQ(bias.size(), output_size);
+  RTC_DCHECK_EQ(weights.size(), input_size * output_size);
+  RTC_DCHECK_EQ(recurrent_weights.size(), output_size * output_size);
+  RTC_DCHECK_GE(gate.size(), output_size);  // `gate` is over-allocated.
+  for (int o = 0; o < output_size; ++o) {
+    float x = bias[o];
+    x += vector_math.DotProduct(input,
+                                weights.subview(o * input_size, input_size));
+    x += vector_math.DotProduct(
+        state, recurrent_weights.subview(o * output_size, output_size));
+    gate[o] = ::rnnoise::SigmoidApproximated(x);
+  }
+}
+
+// Computes the output for the state gate.
+// Operation: `s' = u .* s + (1 - u) .* ReLU(W^T∙i + R^T∙(s .* r) + b)` where
+// - `s'`: output state gate vector
+// - `s`: previous state gate vector
+// - `u`: update gate vector
+// - `W`: weights matrix
+// - `i`: input vector
+// - `R`: recurrent weights matrix
+// - `r`: reset gate vector
+// - `b`: bias vector
+// - `.*` element-wise product
+void ComputeStateGate(int input_size,
+                      int output_size,
+                      const VectorMath& vector_math,
+                      rtc::ArrayView<const float> input,
+                      rtc::ArrayView<const float> update,
+                      rtc::ArrayView<const float> reset,
+                      rtc::ArrayView<const float> bias,
+                      rtc::ArrayView<const float> weights,
+                      rtc::ArrayView<const float> recurrent_weights,
+                      rtc::ArrayView<float> state) {
+  RTC_DCHECK_EQ(input.size(), input_size);
+  RTC_DCHECK_GE(update.size(), output_size);  // `update` is over-allocated.
+  RTC_DCHECK_GE(reset.size(), output_size);   // `reset` is over-allocated.
+  RTC_DCHECK_EQ(bias.size(), output_size);
+  RTC_DCHECK_EQ(weights.size(), input_size * output_size);
+  RTC_DCHECK_EQ(recurrent_weights.size(), output_size * output_size);
+  RTC_DCHECK_EQ(state.size(), output_size);
+  std::array<float, kGruLayerMaxUnits> reset_x_state;
+  for (int o = 0; o < output_size; ++o) {
+    reset_x_state[o] = state[o] * reset[o];
+  }
+  for (int o = 0; o < output_size; ++o) {
+    float x = bias[o];
+    x += vector_math.DotProduct(input,
+                                weights.subview(o * input_size, input_size));
+    x += vector_math.DotProduct(
+        {reset_x_state.data(), static_cast<size_t>(output_size)},
+        recurrent_weights.subview(o * output_size, output_size));
+    state[o] = update[o] * state[o] + (1.f - update[o]) * std::max(0.f, x);
+  }
+}
+
+}  // namespace
+
+GatedRecurrentLayer::GatedRecurrentLayer(
+    const int input_size,
+    const int output_size,
+    const rtc::ArrayView<const int8_t> bias,
+    const rtc::ArrayView<const int8_t> weights,
+    const rtc::ArrayView<const int8_t> recurrent_weights,
+    const AvailableCpuFeatures& cpu_features,
+    absl::string_view layer_name)
+    : input_size_(input_size),
+      output_size_(output_size),
+      bias_(PreprocessGruTensor(bias, output_size)),
+      weights_(PreprocessGruTensor(weights, output_size)),
+      recurrent_weights_(PreprocessGruTensor(recurrent_weights, output_size)),
+      vector_math_(cpu_features) {
+  RTC_DCHECK_LE(output_size_, kGruLayerMaxUnits)
+      << "Insufficient GRU layer over-allocation (" << layer_name << ").";
+  RTC_DCHECK_EQ(kNumGruGates * output_size_, bias_.size())
+      << "Mismatching output size and bias terms array size (" << layer_name
+      << ").";
+  RTC_DCHECK_EQ(kNumGruGates * input_size_ * output_size_, weights_.size())
+      << "Mismatching input-output size and weight coefficients array size ("
+      << layer_name << ").";
+  RTC_DCHECK_EQ(kNumGruGates * output_size_ * output_size_,
+                recurrent_weights_.size())
+      << "Mismatching input-output size and recurrent weight coefficients array"
+         " size ("
+      << layer_name << ").";
+  Reset();
+}
+
+GatedRecurrentLayer::~GatedRecurrentLayer() = default;
+
+void GatedRecurrentLayer::Reset() {
+  state_.fill(0.f);
+}
+
+void GatedRecurrentLayer::ComputeOutput(rtc::ArrayView<const float> input) {
+  RTC_DCHECK_EQ(input.size(), input_size_);
+
+  // The tensors below are organized as a sequence of flattened tensors for the
+  // `update`, `reset` and `state` gates.
+  rtc::ArrayView<const float> bias(bias_);
+  rtc::ArrayView<const float> weights(weights_);
+  rtc::ArrayView<const float> recurrent_weights(recurrent_weights_);
+  // Strides to access to the flattened tensors for a specific gate.
+  const int stride_weights = input_size_ * output_size_;
+  const int stride_recurrent_weights = output_size_ * output_size_;
+
+  rtc::ArrayView<float> state(state_.data(), output_size_);
+
+  // Update gate.
+  std::array<float, kGruLayerMaxUnits> update;
+  ComputeUpdateResetGate(
+      input_size_, output_size_, vector_math_, input, state,
+      bias.subview(0, output_size_), weights.subview(0, stride_weights),
+      recurrent_weights.subview(0, stride_recurrent_weights), update);
+  // Reset gate.
+  std::array<float, kGruLayerMaxUnits> reset;
+  ComputeUpdateResetGate(input_size_, output_size_, vector_math_, input, state,
+                         bias.subview(output_size_, output_size_),
+                         weights.subview(stride_weights, stride_weights),
+                         recurrent_weights.subview(stride_recurrent_weights,
+                                                   stride_recurrent_weights),
+                         reset);
+  // State gate.
+  ComputeStateGate(input_size_, output_size_, vector_math_, input, update,
+                   reset, bias.subview(2 * output_size_, output_size_),
+                   weights.subview(2 * stride_weights, stride_weights),
+                   recurrent_weights.subview(2 * stride_recurrent_weights,
+                                             stride_recurrent_weights),
+                   state);
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
new file mode 100644
index 0000000000..3407dfcdf1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_GRU_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_GRU_H_
+
+#include <array>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Maximum number of units for a GRU layer.
+constexpr int kGruLayerMaxUnits = 24;
+
+// Recurrent layer with gated recurrent units (GRUs) with sigmoid and ReLU as
+// activation functions for the update/reset and output gates respectively.
+class GatedRecurrentLayer {
+ public:
+  // Ctor. `output_size` cannot be greater than `kGruLayerMaxUnits`.
+  GatedRecurrentLayer(int input_size,
+                      int output_size,
+                      rtc::ArrayView<const int8_t> bias,
+                      rtc::ArrayView<const int8_t> weights,
+                      rtc::ArrayView<const int8_t> recurrent_weights,
+                      const AvailableCpuFeatures& cpu_features,
+                      absl::string_view layer_name);
+  GatedRecurrentLayer(const GatedRecurrentLayer&) = delete;
+  GatedRecurrentLayer& operator=(const GatedRecurrentLayer&) = delete;
+  ~GatedRecurrentLayer();
+
+  // Returns the size of the input vector.
+  int input_size() const { return input_size_; }
+  // Returns the pointer to the first element of the output buffer.
+  const float* data() const { return state_.data(); }
+  // Returns the size of the output buffer.
+  int size() const { return output_size_; }
+
+  // Resets the GRU state.
+  void Reset();
+  // Computes the recurrent layer output and updates the status.
+  void ComputeOutput(rtc::ArrayView<const float> input);
+
+ private:
+  const int input_size_;
+  const int output_size_;
+  const std::vector<float> bias_;
+  const std::vector<float> weights_;
+  const std::vector<float> recurrent_weights_;
+  const VectorMath vector_math_;
+  // Over-allocated array with size equal to `output_size_`.
+  std::array<float, kGruLayerMaxUnits> state_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_GRU_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
new file mode 100644
index 0000000000..88ae72803a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2020 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/rnn_vad/rnn_gru.h"
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/test/performance_timer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/gtest.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+void TestGatedRecurrentLayer(
+    GatedRecurrentLayer& gru,
+    rtc::ArrayView<const float> input_sequence,
+    rtc::ArrayView<const float> expected_output_sequence) {
+  const int input_sequence_length = rtc::CheckedDivExact(
+      rtc::dchecked_cast<int>(input_sequence.size()), gru.input_size());
+  const int output_sequence_length = rtc::CheckedDivExact(
+      rtc::dchecked_cast<int>(expected_output_sequence.size()), gru.size());
+  ASSERT_EQ(input_sequence_length, output_sequence_length)
+      << "The test data length is invalid.";
+  // Feed the GRU layer and check the output at every step.
+  gru.Reset();
+  for (int i = 0; i < input_sequence_length; ++i) {
+    SCOPED_TRACE(i);
+    gru.ComputeOutput(
+        input_sequence.subview(i * gru.input_size(), gru.input_size()));
+    const auto expected_output =
+        expected_output_sequence.subview(i * gru.size(), gru.size());
+    ExpectNearAbsolute(expected_output, gru, 3e-6f);
+  }
+}
+
+// Gated recurrent units layer test data.
+constexpr int kGruInputSize = 5;
+constexpr int kGruOutputSize = 4;
+constexpr std::array<int8_t, 12> kGruBias = {96,   -99, -81, -114, 49,  119,
+                                             -118, 68,  -76, 91,   121, 125};
+constexpr std::array<int8_t, 60> kGruWeights = {
+    // Input 0.
+    124, 9, 1, 116,        // Update.
+    -66, -21, -118, -110,  // Reset.
+    104, 75, -23, -51,     // Output.
+    // Input 1.
+    -72, -111, 47, 93,   // Update.
+    77, -98, 41, -8,     // Reset.
+    40, -23, -43, -107,  // Output.
+    // Input 2.
+    9, -73, 30, -32,      // Update.
+    -2, 64, -26, 91,      // Reset.
+    -48, -24, -28, -104,  // Output.
+    // Input 3.
+    74, -46, 116, 15,    // Update.
+    32, 52, -126, -38,   // Reset.
+    -121, 12, -16, 110,  // Output.
+    // Input 4.
+    -95, 66, -103, -35,  // Update.
+    -38, 3, -126, -61,   // Reset.
+    28, 98, -117, -43    // Output.
+};
+constexpr std::array<int8_t, 48> kGruRecurrentWeights = {
+    // Output 0.
+    -3, 87, 50, 51,     // Update.
+    -22, 27, -39, 62,   // Reset.
+    31, -83, -52, -48,  // Output.
+    // Output 1.
+    -6, 83, -19, 104,  // Update.
+    105, 48, 23, 68,   // Reset.
+    23, 40, 7, -120,   // Output.
+    // Output 2.
+    64, -62, 117, 85,     // Update.
+    51, -43, 54, -105,    // Reset.
+    120, 56, -128, -107,  // Output.
+    // Output 3.
+    39, 50, -17, -47,   // Update.
+    -117, 14, 108, 12,  // Reset.
+    -7, -72, 103, -87,  // Output.
+};
+constexpr std::array<float, 20> kGruInputSequence = {
+    0.89395463f, 0.93224651f, 0.55788344f, 0.32341808f, 0.93355054f,
+    0.13475326f, 0.97370994f, 0.14253306f, 0.93710381f, 0.76093364f,
+    0.65780413f, 0.41657975f, 0.49403164f, 0.46843281f, 0.75138855f,
+    0.24517593f, 0.47657707f, 0.57064998f, 0.435184f,   0.19319285f};
+constexpr std::array<float, 16> kGruExpectedOutputSequence = {
+    0.0239123f,  0.5773077f,  0.f,         0.f,
+    0.01282811f, 0.64330572f, 0.f,         0.04863098f,
+    0.00781069f, 0.75267816f, 0.f,         0.02579715f,
+    0.00471378f, 0.59162533f, 0.11087593f, 0.01334511f};
+
+class RnnGruParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+// Checks that the output of a GRU layer is within tolerance given test input
+// data.
+TEST_P(RnnGruParametrization, CheckGatedRecurrentLayer) {
+  GatedRecurrentLayer gru(kGruInputSize, kGruOutputSize, kGruBias, kGruWeights,
+                          kGruRecurrentWeights,
+                          /*cpu_features=*/GetParam(),
+                          /*layer_name=*/"GRU");
+  TestGatedRecurrentLayer(gru, kGruInputSequence, kGruExpectedOutputSequence);
+}
+
+TEST_P(RnnGruParametrization, DISABLED_BenchmarkGatedRecurrentLayer) {
+  // Prefetch test data.
+  std::unique_ptr<FileReader> reader = CreateGruInputReader();
+  std::vector<float> gru_input_sequence(reader->size());
+  reader->ReadChunk(gru_input_sequence);
+
+  using ::rnnoise::kHiddenGruBias;
+  using ::rnnoise::kHiddenGruRecurrentWeights;
+  using ::rnnoise::kHiddenGruWeights;
+  using ::rnnoise::kHiddenLayerOutputSize;
+  using ::rnnoise::kInputLayerOutputSize;
+
+  GatedRecurrentLayer gru(kInputLayerOutputSize, kHiddenLayerOutputSize,
+                          kHiddenGruBias, kHiddenGruWeights,
+                          kHiddenGruRecurrentWeights,
+                          /*cpu_features=*/GetParam(),
+                          /*layer_name=*/"GRU");
+
+  rtc::ArrayView<const float> input_sequence(gru_input_sequence);
+  ASSERT_EQ(input_sequence.size() % kInputLayerOutputSize,
+            static_cast<size_t>(0));
+  const int input_sequence_length =
+      input_sequence.size() / kInputLayerOutputSize;
+
+  constexpr int kNumTests = 100;
+  ::webrtc::test::PerformanceTimer perf_timer(kNumTests);
+  for (int k = 0; k < kNumTests; ++k) {
+    perf_timer.StartTimer();
+    for (int i = 0; i < input_sequence_length; ++i) {
+      gru.ComputeOutput(
+          input_sequence.subview(i * gru.input_size(), gru.input_size()));
+    }
+    perf_timer.StopTimer();
+  }
+  RTC_LOG(LS_INFO) << (perf_timer.GetDurationAverage() / 1000) << " +/- "
+                   << (perf_timer.GetDurationStandardDeviation() / 1000)
+                   << " ms";
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    RnnGruParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
new file mode 100644
index 0000000000..4c5409a14e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
@@ -0,0 +1,70 @@
+/*
+ *  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/rnn_vad/rnn.h"
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr std::array<float, kFeatureVectorSize> kFeatures = {
+    -1.00131f,   -0.627069f, -7.81097f,  7.86285f,    -2.87145f,  3.32365f,
+    -0.653161f,  0.529839f,  -0.425307f, 0.25583f,    0.235094f,  0.230527f,
+    -0.144687f,  0.182785f,  0.57102f,   0.125039f,   0.479482f,  -0.0255439f,
+    -0.0073141f, -0.147346f, -0.217106f, -0.0846906f, -8.34943f,  3.09065f,
+    1.42628f,    -0.85235f,  -0.220207f, -0.811163f,  2.09032f,   -2.01425f,
+    -0.690268f,  -0.925327f, -0.541354f, 0.58455f,    -0.606726f, -0.0372358f,
+    0.565991f,   0.435854f,  0.420812f,  0.162198f,   -2.13f,     10.0089f};
+
+void WarmUpRnnVad(RnnVad& rnn_vad) {
+  for (int i = 0; i < 10; ++i) {
+    rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  }
+}
+
+// Checks that the speech probability is zero with silence.
+TEST(RnnVadTest, CheckZeroProbabilityWithSilence) {
+  RnnVad rnn_vad(GetAvailableCpuFeatures());
+  WarmUpRnnVad(rnn_vad);
+  EXPECT_EQ(rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/true), 0.f);
+}
+
+// Checks that the same output is produced after reset given the same input
+// sequence.
+TEST(RnnVadTest, CheckRnnVadReset) {
+  RnnVad rnn_vad(GetAvailableCpuFeatures());
+  WarmUpRnnVad(rnn_vad);
+  float pre = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  rnn_vad.Reset();
+  WarmUpRnnVad(rnn_vad);
+  float post = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  EXPECT_EQ(pre, post);
+}
+
+// Checks that the same output is produced after silence is observed given the
+// same input sequence.
+TEST(RnnVadTest, CheckRnnVadSilence) {
+  RnnVad rnn_vad(GetAvailableCpuFeatures());
+  WarmUpRnnVad(rnn_vad);
+  float pre = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/true);
+  WarmUpRnnVad(rnn_vad);
+  float post = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  EXPECT_EQ(pre, post);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build
new file mode 100644
index 0000000000..134ffac5fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build
@@ -0,0 +1,236 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_auto_correlation_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build
new file mode 100644
index 0000000000..cf3de48a57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build
@@ -0,0 +1,220 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_common_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build
new file mode 100644
index 0000000000..dbb926c5fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build
@@ -0,0 +1,237 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build
new file mode 100644
index 0000000000..92da260f90
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build
@@ -0,0 +1,237 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_layers_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build
new file mode 100644
index 0000000000..cedb17bc22
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_lp_residual_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build
new file mode 100644
index 0000000000..d45bc78ff6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build
@@ -0,0 +1,237 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_pitch_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build
new file mode 100644
index 0000000000..20da5f3615
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build
@@ -0,0 +1,209 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_ring_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build
new file mode 100644
index 0000000000..b0ba79562e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build
@@ -0,0 +1,209 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_sequence_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build
new file mode 100644
index 0000000000..2d8396fa2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build
@@ -0,0 +1,237 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_spectral_features_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build
new file mode 100644
index 0000000000..143ba6960c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build
@@ -0,0 +1,209 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_symmetric_matrix_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
new file mode 100644
index 0000000000..a0e1242eb4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
@@ -0,0 +1,123 @@
+/*
+ *  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 <array>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+ABSL_FLAG(std::string, i, "", "Path to the input wav file");
+ABSL_FLAG(std::string, f, "", "Path to the output features file");
+ABSL_FLAG(std::string, o, "", "Path to the output VAD probabilities file");
+
+namespace webrtc {
+namespace rnn_vad {
+namespace test {
+
+int main(int argc, char* argv[]) {
+  absl::ParseCommandLine(argc, argv);
+  rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+
+  // Open wav input file and check properties.
+  const std::string input_wav_file = absl::GetFlag(FLAGS_i);
+  WavReader wav_reader(input_wav_file);
+  if (wav_reader.num_channels() != 1) {
+    RTC_LOG(LS_ERROR) << "Only mono wav files are supported";
+    return 1;
+  }
+  if (wav_reader.sample_rate() % 100 != 0) {
+    RTC_LOG(LS_ERROR) << "The sample rate rate must allow 10 ms frames.";
+    return 1;
+  }
+  RTC_LOG(LS_INFO) << "Input sample rate: " << wav_reader.sample_rate();
+
+  // Init output files.
+  const std::string output_vad_probs_file = absl::GetFlag(FLAGS_o);
+  FILE* vad_probs_file = fopen(output_vad_probs_file.c_str(), "wb");
+  FILE* features_file = nullptr;
+  const std::string output_feature_file = absl::GetFlag(FLAGS_f);
+  if (!output_feature_file.empty()) {
+    features_file = fopen(output_feature_file.c_str(), "wb");
+  }
+
+  // Initialize.
+  const int frame_size_10ms =
+      rtc::CheckedDivExact(wav_reader.sample_rate(), 100);
+  std::vector<float> samples_10ms;
+  samples_10ms.resize(frame_size_10ms);
+  std::array<float, kFrameSize10ms24kHz> samples_10ms_24kHz;
+  PushSincResampler resampler(frame_size_10ms, kFrameSize10ms24kHz);
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+  FeaturesExtractor features_extractor(cpu_features);
+  std::array<float, kFeatureVectorSize> feature_vector;
+  RnnVad rnn_vad(cpu_features);
+
+  // Compute VAD probabilities.
+  while (true) {
+    // Read frame at the input sample rate.
+    const size_t read_samples =
+        wav_reader.ReadSamples(frame_size_10ms, samples_10ms.data());
+    if (rtc::SafeLt(read_samples, frame_size_10ms)) {
+      break;  // EOF.
+    }
+    // Resample input.
+    resampler.Resample(samples_10ms.data(), samples_10ms.size(),
+                       samples_10ms_24kHz.data(), samples_10ms_24kHz.size());
+
+    // Extract features and feed the RNN.
+    bool is_silence = features_extractor.CheckSilenceComputeFeatures(
+        samples_10ms_24kHz, feature_vector);
+    float vad_probability =
+        rnn_vad.ComputeVadProbability(feature_vector, is_silence);
+    // Write voice probability.
+    RTC_DCHECK_GE(vad_probability, 0.f);
+    RTC_DCHECK_GE(1.f, vad_probability);
+    fwrite(&vad_probability, sizeof(float), 1, vad_probs_file);
+    // Write features.
+    if (features_file) {
+      const float float_is_silence = is_silence ? 1.f : 0.f;
+      fwrite(&float_is_silence, sizeof(float), 1, features_file);
+      if (is_silence) {
+        // Do not write uninitialized values.
+        feature_vector.fill(0.f);
+      }
+      fwrite(feature_vector.data(), sizeof(float), kFeatureVectorSize,
+             features_file);
+    }
+  }
+
+  // Close output file(s).
+  fclose(vad_probs_file);
+  RTC_LOG(LS_INFO) << "VAD probabilities written to " << output_vad_probs_file;
+  if (features_file) {
+    fclose(features_file);
+    RTC_LOG(LS_INFO) << "features written to " << output_feature_file;
+  }
+
+  return 0;
+}
+
+}  // namespace test
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::rnn_vad::test::main(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
new file mode 100644
index 0000000000..f33cd14a8a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
@@ -0,0 +1,185 @@
+/*
+ *  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 <array>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/test/performance_timer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/gtest.h"
+#include "third_party/rnnoise/src/rnn_activations.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kFrameSize10ms48kHz = 480;
+
+void DumpPerfStats(int num_samples,
+                   int sample_rate,
+                   double average_us,
+                   double standard_deviation) {
+  float audio_track_length_ms =
+      1e3f * static_cast<float>(num_samples) / static_cast<float>(sample_rate);
+  float average_ms = static_cast<float>(average_us) / 1e3f;
+  float speed = audio_track_length_ms / average_ms;
+  RTC_LOG(LS_INFO) << "track duration (ms): " << audio_track_length_ms;
+  RTC_LOG(LS_INFO) << "average processing time (ms): " << average_ms << " +/- "
+                   << (standard_deviation / 1e3);
+  RTC_LOG(LS_INFO) << "speed: " << speed << "x";
+}
+
+// When the RNN VAD model is updated and the expected output changes, set the
+// constant below to true in order to write new expected output binary files.
+constexpr bool kWriteComputedOutputToFile = false;
+
+// Avoids that one forgets to set `kWriteComputedOutputToFile` back to false
+// when the expected output files are re-exported.
+TEST(RnnVadTest, CheckWriteComputedOutputIsFalse) {
+  ASSERT_FALSE(kWriteComputedOutputToFile)
+      << "Cannot land if kWriteComputedOutput is true.";
+}
+
+class RnnVadProbabilityParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+// Checks that the computed VAD probability for a test input sequence sampled at
+// 48 kHz is within tolerance.
+TEST_P(RnnVadProbabilityParametrization, RnnVadProbabilityWithinTolerance) {
+  // Init resampler, feature extractor and RNN.
+  PushSincResampler decimator(kFrameSize10ms48kHz, kFrameSize10ms24kHz);
+  const AvailableCpuFeatures cpu_features = GetParam();
+  FeaturesExtractor features_extractor(cpu_features);
+  RnnVad rnn_vad(cpu_features);
+
+  // Init input samples and expected output readers.
+  std::unique_ptr<FileReader> samples_reader = CreatePcmSamplesReader();
+  std::unique_ptr<FileReader> expected_vad_prob_reader = CreateVadProbsReader();
+
+  // Input length. The last incomplete frame is ignored.
+  const int num_frames = samples_reader->size() / kFrameSize10ms48kHz;
+
+  // Init buffers.
+  std::vector<float> samples_48k(kFrameSize10ms48kHz);
+  std::vector<float> samples_24k(kFrameSize10ms24kHz);
+  std::vector<float> feature_vector(kFeatureVectorSize);
+  std::vector<float> computed_vad_prob(num_frames);
+  std::vector<float> expected_vad_prob(num_frames);
+
+  // Read expected output.
+  ASSERT_TRUE(expected_vad_prob_reader->ReadChunk(expected_vad_prob));
+
+  // Compute VAD probabilities on the downsampled input.
+  float cumulative_error = 0.f;
+  for (int i = 0; i < num_frames; ++i) {
+    ASSERT_TRUE(samples_reader->ReadChunk(samples_48k));
+    decimator.Resample(samples_48k.data(), samples_48k.size(),
+                       samples_24k.data(), samples_24k.size());
+    bool is_silence = features_extractor.CheckSilenceComputeFeatures(
+        {samples_24k.data(), kFrameSize10ms24kHz},
+        {feature_vector.data(), kFeatureVectorSize});
+    computed_vad_prob[i] = rnn_vad.ComputeVadProbability(
+        {feature_vector.data(), kFeatureVectorSize}, is_silence);
+    EXPECT_NEAR(computed_vad_prob[i], expected_vad_prob[i], 1e-3f);
+    cumulative_error += std::abs(computed_vad_prob[i] - expected_vad_prob[i]);
+  }
+  // Check average error.
+  EXPECT_LT(cumulative_error / num_frames, 1e-4f);
+
+  if (kWriteComputedOutputToFile) {
+    FileWriter vad_prob_writer("new_vad_prob.dat");
+    vad_prob_writer.WriteChunk(computed_vad_prob);
+  }
+}
+
+// Performance test for the RNN VAD (pre-fetching and downsampling are
+// excluded). Keep disabled and only enable locally to measure performance as
+// follows:
+// - on desktop: run the this unit test adding "--logs";
+// - on android: run the this unit test adding "--logcat-output-file".
+TEST_P(RnnVadProbabilityParametrization, DISABLED_RnnVadPerformance) {
+  // PCM samples reader and buffers.
+  std::unique_ptr<FileReader> samples_reader = CreatePcmSamplesReader();
+  // The last incomplete frame is ignored.
+  const int num_frames = samples_reader->size() / kFrameSize10ms48kHz;
+  std::array<float, kFrameSize10ms48kHz> samples;
+  // Pre-fetch and decimate samples.
+  PushSincResampler decimator(kFrameSize10ms48kHz, kFrameSize10ms24kHz);
+  std::vector<float> prefetched_decimated_samples;
+  prefetched_decimated_samples.resize(num_frames * kFrameSize10ms24kHz);
+  for (int i = 0; i < num_frames; ++i) {
+    ASSERT_TRUE(samples_reader->ReadChunk(samples));
+    decimator.Resample(samples.data(), samples.size(),
+                       &prefetched_decimated_samples[i * kFrameSize10ms24kHz],
+                       kFrameSize10ms24kHz);
+  }
+  // Initialize.
+  const AvailableCpuFeatures cpu_features = GetParam();
+  FeaturesExtractor features_extractor(cpu_features);
+  std::array<float, kFeatureVectorSize> feature_vector;
+  RnnVad rnn_vad(cpu_features);
+  constexpr int number_of_tests = 100;
+  ::webrtc::test::PerformanceTimer perf_timer(number_of_tests);
+  for (int k = 0; k < number_of_tests; ++k) {
+    features_extractor.Reset();
+    rnn_vad.Reset();
+    // Process frames.
+    perf_timer.StartTimer();
+    for (int i = 0; i < num_frames; ++i) {
+      bool is_silence = features_extractor.CheckSilenceComputeFeatures(
+          {&prefetched_decimated_samples[i * kFrameSize10ms24kHz],
+           kFrameSize10ms24kHz},
+          feature_vector);
+      rnn_vad.ComputeVadProbability(feature_vector, is_silence);
+    }
+    perf_timer.StopTimer();
+  }
+  DumpPerfStats(num_frames * kFrameSize10ms24kHz, kSampleRate24kHz,
+                perf_timer.GetDurationAverage(),
+                perf_timer.GetDurationStandardDeviation());
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.avx2 && available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    RnnVadProbabilityParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
new file mode 100644
index 0000000000..a7402788c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
@@ -0,0 +1,79 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
+
+#include <algorithm>
+#include <cstring>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Linear buffer implementation to (i) push fixed size chunks of sequential data
+// and (ii) view contiguous parts of the buffer. The buffer and the pushed
+// chunks have size S and N respectively. For instance, when S = 2N the first
+// half of the sequence buffer is replaced with its second half, and the new N
+// values are written at the end of the buffer.
+// The class also provides a view on the most recent M values, where 0 < M <= S
+// and by default M = N.
+template <typename T, int S, int N, int M = N>
+class SequenceBuffer {
+  static_assert(N <= S,
+                "The new chunk size cannot be larger than the sequence buffer "
+                "size.");
+  static_assert(std::is_arithmetic<T>::value,
+                "Integral or floating point required.");
+
+ public:
+  SequenceBuffer() : buffer_(S) {
+    RTC_DCHECK_EQ(S, buffer_.size());
+    Reset();
+  }
+  SequenceBuffer(const SequenceBuffer&) = delete;
+  SequenceBuffer& operator=(const SequenceBuffer&) = delete;
+  ~SequenceBuffer() = default;
+  int size() const { return S; }
+  int chunks_size() const { return N; }
+  // Sets the sequence buffer values to zero.
+  void Reset() { std::fill(buffer_.begin(), buffer_.end(), 0); }
+  // Returns a view on the whole buffer.
+  rtc::ArrayView<const T, S> GetBufferView() const {
+    return {buffer_.data(), S};
+  }
+  // Returns a view on the M most recent values of the buffer.
+  rtc::ArrayView<const T, M> GetMostRecentValuesView() const {
+    static_assert(M <= S,
+                  "The number of most recent values cannot be larger than the "
+                  "sequence buffer size.");
+    return {buffer_.data() + S - M, M};
+  }
+  // Shifts left the buffer by N items and add new N items at the end.
+  void Push(rtc::ArrayView<const T, N> new_values) {
+    // Make space for the new values.
+    if (S > N)
+      std::memmove(buffer_.data(), buffer_.data() + N, (S - N) * sizeof(T));
+    // Copy the new values at the end of the buffer.
+    std::memcpy(buffer_.data() + S - N, new_values.data(), N * sizeof(T));
+  }
+
+ private:
+  std::vector<T> buffer_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
new file mode 100644
index 0000000000..af005833c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
@@ -0,0 +1,102 @@
+/*
+ *  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/rnn_vad/sequence_buffer.h"
+
+#include <algorithm>
+#include <array>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+template <typename T, int S, int N>
+void TestSequenceBufferPushOp() {
+  SCOPED_TRACE(S);
+  SCOPED_TRACE(N);
+  SequenceBuffer<T, S, N> seq_buf;
+  auto seq_buf_view = seq_buf.GetBufferView();
+  std::array<T, N> chunk;
+
+  // Check that a chunk is fully gone after ceil(S / N) push ops.
+  chunk.fill(1);
+  seq_buf.Push(chunk);
+  chunk.fill(0);
+  constexpr int required_push_ops = (S % N) ? S / N + 1 : S / N;
+  for (int i = 0; i < required_push_ops - 1; ++i) {
+    SCOPED_TRACE(i);
+    seq_buf.Push(chunk);
+    // Still in the buffer.
+    const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
+    EXPECT_EQ(1, *m);
+  }
+  // Gone after another push.
+  seq_buf.Push(chunk);
+  const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
+  EXPECT_EQ(0, *m);
+
+  // Check that the last item moves left by N positions after a push op.
+  if (S > N) {
+    // Fill in with non-zero values.
+    for (int i = 0; i < N; ++i)
+      chunk[i] = static_cast<T>(i + 1);
+    seq_buf.Push(chunk);
+    // With the next Push(), `last` will be moved left by N positions.
+    const T last = chunk[N - 1];
+    for (int i = 0; i < N; ++i)
+      chunk[i] = static_cast<T>(last + i + 1);
+    seq_buf.Push(chunk);
+    EXPECT_EQ(last, seq_buf_view[S - N - 1]);
+  }
+}
+
+TEST(RnnVadTest, SequenceBufferGetters) {
+  constexpr int buffer_size = 8;
+  constexpr int chunk_size = 8;
+  SequenceBuffer<int, buffer_size, chunk_size> seq_buf;
+  EXPECT_EQ(buffer_size, seq_buf.size());
+  EXPECT_EQ(chunk_size, seq_buf.chunks_size());
+  // Test view.
+  auto seq_buf_view = seq_buf.GetBufferView();
+  EXPECT_EQ(0, seq_buf_view[0]);
+  EXPECT_EQ(0, seq_buf_view[seq_buf_view.size() - 1]);
+  constexpr std::array<int, chunk_size> chunk = {10, 20, 30, 40,
+                                                 50, 60, 70, 80};
+  seq_buf.Push(chunk);
+  EXPECT_EQ(10, *seq_buf_view.begin());
+  EXPECT_EQ(80, *(seq_buf_view.end() - 1));
+}
+
+TEST(RnnVadTest, SequenceBufferPushOpsUnsigned) {
+  TestSequenceBufferPushOp<uint8_t, 32, 8>();   // Chunk size: 25%.
+  TestSequenceBufferPushOp<uint8_t, 32, 16>();  // Chunk size: 50%.
+  TestSequenceBufferPushOp<uint8_t, 32, 32>();  // Chunk size: 100%.
+  TestSequenceBufferPushOp<uint8_t, 23, 7>();   // Non-integer ratio.
+}
+
+TEST(RnnVadTest, SequenceBufferPushOpsSigned) {
+  TestSequenceBufferPushOp<int, 32, 8>();   // Chunk size: 25%.
+  TestSequenceBufferPushOp<int, 32, 16>();  // Chunk size: 50%.
+  TestSequenceBufferPushOp<int, 32, 32>();  // Chunk size: 100%.
+  TestSequenceBufferPushOp<int, 23, 7>();   // Non-integer ratio.
+}
+
+TEST(RnnVadTest, SequenceBufferPushOpsFloating) {
+  TestSequenceBufferPushOp<float, 32, 8>();   // Chunk size: 25%.
+  TestSequenceBufferPushOp<float, 32, 16>();  // Chunk size: 50%.
+  TestSequenceBufferPushOp<float, 32, 32>();  // Chunk size: 100%.
+  TestSequenceBufferPushOp<float, 23, 7>();   // Non-integer ratio.
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
new file mode 100644
index 0000000000..96086babb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
@@ -0,0 +1,214 @@
+/*
+ *  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/rnn_vad/spectral_features.h"
+
+#include <algorithm>
+#include <cmath>
+#include <limits>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr float kSilenceThreshold = 0.04f;
+
+// Computes the new cepstral difference stats and pushes them into the passed
+// symmetric matrix buffer.
+void UpdateCepstralDifferenceStats(
+    rtc::ArrayView<const float, kNumBands> new_cepstral_coeffs,
+    const RingBuffer<float, kNumBands, kCepstralCoeffsHistorySize>& ring_buf,
+    SymmetricMatrixBuffer<float, kCepstralCoeffsHistorySize>* sym_matrix_buf) {
+  RTC_DCHECK(sym_matrix_buf);
+  // Compute the new cepstral distance stats.
+  std::array<float, kCepstralCoeffsHistorySize - 1> distances;
+  for (int i = 0; i < kCepstralCoeffsHistorySize - 1; ++i) {
+    const int delay = i + 1;
+    auto old_cepstral_coeffs = ring_buf.GetArrayView(delay);
+    distances[i] = 0.f;
+    for (int k = 0; k < kNumBands; ++k) {
+      const float c = new_cepstral_coeffs[k] - old_cepstral_coeffs[k];
+      distances[i] += c * c;
+    }
+  }
+  // Push the new spectral distance stats into the symmetric matrix buffer.
+  sym_matrix_buf->Push(distances);
+}
+
+// Computes the first half of the Vorbis window.
+std::array<float, kFrameSize20ms24kHz / 2> ComputeScaledHalfVorbisWindow(
+    float scaling = 1.f) {
+  constexpr int kHalfSize = kFrameSize20ms24kHz / 2;
+  std::array<float, kHalfSize> half_window{};
+  for (int i = 0; i < kHalfSize; ++i) {
+    half_window[i] =
+        scaling *
+        std::sin(0.5 * kPi * std::sin(0.5 * kPi * (i + 0.5) / kHalfSize) *
+                 std::sin(0.5 * kPi * (i + 0.5) / kHalfSize));
+  }
+  return half_window;
+}
+
+// Computes the forward FFT on a 20 ms frame to which a given window function is
+// applied. The Fourier coefficient corresponding to the Nyquist frequency is
+// set to zero (it is never used and this allows to simplify the code).
+void ComputeWindowedForwardFft(
+    rtc::ArrayView<const float, kFrameSize20ms24kHz> frame,
+    const std::array<float, kFrameSize20ms24kHz / 2>& half_window,
+    Pffft::FloatBuffer* fft_input_buffer,
+    Pffft::FloatBuffer* fft_output_buffer,
+    Pffft* fft) {
+  RTC_DCHECK_EQ(frame.size(), 2 * half_window.size());
+  // Apply windowing.
+  auto in = fft_input_buffer->GetView();
+  for (int i = 0, j = kFrameSize20ms24kHz - 1;
+       rtc::SafeLt(i, half_window.size()); ++i, --j) {
+    in[i] = frame[i] * half_window[i];
+    in[j] = frame[j] * half_window[i];
+  }
+  fft->ForwardTransform(*fft_input_buffer, fft_output_buffer, /*ordered=*/true);
+  // Set the Nyquist frequency coefficient to zero.
+  auto out = fft_output_buffer->GetView();
+  out[1] = 0.f;
+}
+
+}  // namespace
+
+SpectralFeaturesExtractor::SpectralFeaturesExtractor()
+    : half_window_(ComputeScaledHalfVorbisWindow(
+          1.f / static_cast<float>(kFrameSize20ms24kHz))),
+      fft_(kFrameSize20ms24kHz, Pffft::FftType::kReal),
+      fft_buffer_(fft_.CreateBuffer()),
+      reference_frame_fft_(fft_.CreateBuffer()),
+      lagged_frame_fft_(fft_.CreateBuffer()),
+      dct_table_(ComputeDctTable()) {}
+
+SpectralFeaturesExtractor::~SpectralFeaturesExtractor() = default;
+
+void SpectralFeaturesExtractor::Reset() {
+  cepstral_coeffs_ring_buf_.Reset();
+  cepstral_diffs_buf_.Reset();
+}
+
+bool SpectralFeaturesExtractor::CheckSilenceComputeFeatures(
+    rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
+    rtc::ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
+    rtc::ArrayView<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
+    rtc::ArrayView<float, kNumLowerBands> average,
+    rtc::ArrayView<float, kNumLowerBands> first_derivative,
+    rtc::ArrayView<float, kNumLowerBands> second_derivative,
+    rtc::ArrayView<float, kNumLowerBands> bands_cross_corr,
+    float* variability) {
+  // Compute the Opus band energies for the reference frame.
+  ComputeWindowedForwardFft(reference_frame, half_window_, fft_buffer_.get(),
+                            reference_frame_fft_.get(), &fft_);
+  spectral_correlator_.ComputeAutoCorrelation(
+      reference_frame_fft_->GetConstView(), reference_frame_bands_energy_);
+  // Check if the reference frame has silence.
+  const float tot_energy =
+      std::accumulate(reference_frame_bands_energy_.begin(),
+                      reference_frame_bands_energy_.end(), 0.f);
+  if (tot_energy < kSilenceThreshold) {
+    return true;
+  }
+  // Compute the Opus band energies for the lagged frame.
+  ComputeWindowedForwardFft(lagged_frame, half_window_, fft_buffer_.get(),
+                            lagged_frame_fft_.get(), &fft_);
+  spectral_correlator_.ComputeAutoCorrelation(lagged_frame_fft_->GetConstView(),
+                                              lagged_frame_bands_energy_);
+  // Log of the band energies for the reference frame.
+  std::array<float, kNumBands> log_bands_energy;
+  ComputeSmoothedLogMagnitudeSpectrum(reference_frame_bands_energy_,
+                                      log_bands_energy);
+  // Reference frame cepstrum.
+  std::array<float, kNumBands> cepstrum;
+  ComputeDct(log_bands_energy, dct_table_, cepstrum);
+  // Ad-hoc correction terms for the first two cepstral coefficients.
+  cepstrum[0] -= 12.f;
+  cepstrum[1] -= 4.f;
+  // Update the ring buffer and the cepstral difference stats.
+  cepstral_coeffs_ring_buf_.Push(cepstrum);
+  UpdateCepstralDifferenceStats(cepstrum, cepstral_coeffs_ring_buf_,
+                                &cepstral_diffs_buf_);
+  // Write the higher bands cepstral coefficients.
+  RTC_DCHECK_EQ(cepstrum.size() - kNumLowerBands, higher_bands_cepstrum.size());
+  std::copy(cepstrum.begin() + kNumLowerBands, cepstrum.end(),
+            higher_bands_cepstrum.begin());
+  // Compute and write remaining features.
+  ComputeAvgAndDerivatives(average, first_derivative, second_derivative);
+  ComputeNormalizedCepstralCorrelation(bands_cross_corr);
+  RTC_DCHECK(variability);
+  *variability = ComputeVariability();
+  return false;
+}
+
+void SpectralFeaturesExtractor::ComputeAvgAndDerivatives(
+    rtc::ArrayView<float, kNumLowerBands> average,
+    rtc::ArrayView<float, kNumLowerBands> first_derivative,
+    rtc::ArrayView<float, kNumLowerBands> second_derivative) const {
+  auto curr = cepstral_coeffs_ring_buf_.GetArrayView(0);
+  auto prev1 = cepstral_coeffs_ring_buf_.GetArrayView(1);
+  auto prev2 = cepstral_coeffs_ring_buf_.GetArrayView(2);
+  RTC_DCHECK_EQ(average.size(), first_derivative.size());
+  RTC_DCHECK_EQ(first_derivative.size(), second_derivative.size());
+  RTC_DCHECK_LE(average.size(), curr.size());
+  for (int i = 0; rtc::SafeLt(i, average.size()); ++i) {
+    // Average, kernel: [1, 1, 1].
+    average[i] = curr[i] + prev1[i] + prev2[i];
+    // First derivative, kernel: [1, 0, - 1].
+    first_derivative[i] = curr[i] - prev2[i];
+    // Second derivative, Laplacian kernel: [1, -2, 1].
+    second_derivative[i] = curr[i] - 2 * prev1[i] + prev2[i];
+  }
+}
+
+void SpectralFeaturesExtractor::ComputeNormalizedCepstralCorrelation(
+    rtc::ArrayView<float, kNumLowerBands> bands_cross_corr) {
+  spectral_correlator_.ComputeCrossCorrelation(
+      reference_frame_fft_->GetConstView(), lagged_frame_fft_->GetConstView(),
+      bands_cross_corr_);
+  // Normalize.
+  for (int i = 0; rtc::SafeLt(i, bands_cross_corr_.size()); ++i) {
+    bands_cross_corr_[i] =
+        bands_cross_corr_[i] /
+        std::sqrt(0.001f + reference_frame_bands_energy_[i] *
+                               lagged_frame_bands_energy_[i]);
+  }
+  // Cepstrum.
+  ComputeDct(bands_cross_corr_, dct_table_, bands_cross_corr);
+  // Ad-hoc correction terms for the first two cepstral coefficients.
+  bands_cross_corr[0] -= 1.3f;
+  bands_cross_corr[1] -= 0.9f;
+}
+
+float SpectralFeaturesExtractor::ComputeVariability() const {
+  // Compute cepstral variability score.
+  float variability = 0.f;
+  for (int delay1 = 0; delay1 < kCepstralCoeffsHistorySize; ++delay1) {
+    float min_dist = std::numeric_limits<float>::max();
+    for (int delay2 = 0; delay2 < kCepstralCoeffsHistorySize; ++delay2) {
+      if (delay1 == delay2)  // The distance would be 0.
+        continue;
+      min_dist =
+          std::min(min_dist, cepstral_diffs_buf_.GetValue(delay1, delay2));
+    }
+    variability += min_dist;
+  }
+  // Normalize (based on training set stats).
+  // TODO(bugs.webrtc.org/10480): Isolate normalization from feature extraction.
+  return variability / kCepstralCoeffsHistorySize - 2.1f;
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
new file mode 100644
index 0000000000..d327ef8e01
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
@@ -0,0 +1,79 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
+
+#include <array>
+#include <cstddef>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
+#include "modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h"
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Class to compute spectral features.
+class SpectralFeaturesExtractor {
+ public:
+  SpectralFeaturesExtractor();
+  SpectralFeaturesExtractor(const SpectralFeaturesExtractor&) = delete;
+  SpectralFeaturesExtractor& operator=(const SpectralFeaturesExtractor&) =
+      delete;
+  ~SpectralFeaturesExtractor();
+  // Resets the internal state of the feature extractor.
+  void Reset();
+  // Analyzes a pair of reference and lagged frames from the pitch buffer,
+  // detects silence and computes features. If silence is detected, the output
+  // is neither computed nor written.
+  bool CheckSilenceComputeFeatures(
+      rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
+      rtc::ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
+      rtc::ArrayView<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
+      rtc::ArrayView<float, kNumLowerBands> average,
+      rtc::ArrayView<float, kNumLowerBands> first_derivative,
+      rtc::ArrayView<float, kNumLowerBands> second_derivative,
+      rtc::ArrayView<float, kNumLowerBands> bands_cross_corr,
+      float* variability);
+
+ private:
+  void ComputeAvgAndDerivatives(
+      rtc::ArrayView<float, kNumLowerBands> average,
+      rtc::ArrayView<float, kNumLowerBands> first_derivative,
+      rtc::ArrayView<float, kNumLowerBands> second_derivative) const;
+  void ComputeNormalizedCepstralCorrelation(
+      rtc::ArrayView<float, kNumLowerBands> bands_cross_corr);
+  float ComputeVariability() const;
+
+  const std::array<float, kFrameSize20ms24kHz / 2> half_window_;
+  Pffft fft_;
+  std::unique_ptr<Pffft::FloatBuffer> fft_buffer_;
+  std::unique_ptr<Pffft::FloatBuffer> reference_frame_fft_;
+  std::unique_ptr<Pffft::FloatBuffer> lagged_frame_fft_;
+  SpectralCorrelator spectral_correlator_;
+  std::array<float, kOpusBands24kHz> reference_frame_bands_energy_;
+  std::array<float, kOpusBands24kHz> lagged_frame_bands_energy_;
+  std::array<float, kOpusBands24kHz> bands_cross_corr_;
+  const std::array<float, kNumBands * kNumBands> dct_table_;
+  RingBuffer<float, kNumBands, kCepstralCoeffsHistorySize>
+      cepstral_coeffs_ring_buf_;
+  SymmetricMatrixBuffer<float, kCepstralCoeffsHistorySize> cepstral_diffs_buf_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
new file mode 100644
index 0000000000..a10b0f7ec9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
@@ -0,0 +1,188 @@
+/*
+ *  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/rnn_vad/spectral_features_internal.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Weights for each FFT coefficient for each Opus band (Nyquist frequency
+// excluded). The size of each band is specified in
+// `kOpusScaleNumBins24kHz20ms`.
+constexpr std::array<float, kFrameSize20ms24kHz / 2> kOpusBandWeights24kHz20ms =
+    {{
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 0
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 1
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 2
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 3
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 4
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 5
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 6
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 7
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 8
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 9
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 10
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 11
+        0.f,       0.0625f,    0.125f,     0.1875f,   0.25f,
+        0.3125f,   0.375f,     0.4375f,    0.5f,      0.5625f,
+        0.625f,    0.6875f,    0.75f,      0.8125f,   0.875f,
+        0.9375f,  // Band 12
+        0.f,       0.0625f,    0.125f,     0.1875f,   0.25f,
+        0.3125f,   0.375f,     0.4375f,    0.5f,      0.5625f,
+        0.625f,    0.6875f,    0.75f,      0.8125f,   0.875f,
+        0.9375f,  // Band 13
+        0.f,       0.0625f,    0.125f,     0.1875f,   0.25f,
+        0.3125f,   0.375f,     0.4375f,    0.5f,      0.5625f,
+        0.625f,    0.6875f,    0.75f,      0.8125f,   0.875f,
+        0.9375f,  // Band 14
+        0.f,       0.0416667f, 0.0833333f, 0.125f,    0.166667f,
+        0.208333f, 0.25f,      0.291667f,  0.333333f, 0.375f,
+        0.416667f, 0.458333f,  0.5f,       0.541667f, 0.583333f,
+        0.625f,    0.666667f,  0.708333f,  0.75f,     0.791667f,
+        0.833333f, 0.875f,     0.916667f,  0.958333f,  // Band 15
+        0.f,       0.0416667f, 0.0833333f, 0.125f,    0.166667f,
+        0.208333f, 0.25f,      0.291667f,  0.333333f, 0.375f,
+        0.416667f, 0.458333f,  0.5f,       0.541667f, 0.583333f,
+        0.625f,    0.666667f,  0.708333f,  0.75f,     0.791667f,
+        0.833333f, 0.875f,     0.916667f,  0.958333f,  // Band 16
+        0.f,       0.03125f,   0.0625f,    0.09375f,  0.125f,
+        0.15625f,  0.1875f,    0.21875f,   0.25f,     0.28125f,
+        0.3125f,   0.34375f,   0.375f,     0.40625f,  0.4375f,
+        0.46875f,  0.5f,       0.53125f,   0.5625f,   0.59375f,
+        0.625f,    0.65625f,   0.6875f,    0.71875f,  0.75f,
+        0.78125f,  0.8125f,    0.84375f,   0.875f,    0.90625f,
+        0.9375f,   0.96875f,  // Band 17
+        0.f,       0.0208333f, 0.0416667f, 0.0625f,   0.0833333f,
+        0.104167f, 0.125f,     0.145833f,  0.166667f, 0.1875f,
+        0.208333f, 0.229167f,  0.25f,      0.270833f, 0.291667f,
+        0.3125f,   0.333333f,  0.354167f,  0.375f,    0.395833f,
+        0.416667f, 0.4375f,    0.458333f,  0.479167f, 0.5f,
+        0.520833f, 0.541667f,  0.5625f,    0.583333f, 0.604167f,
+        0.625f,    0.645833f,  0.666667f,  0.6875f,   0.708333f,
+        0.729167f, 0.75f,      0.770833f,  0.791667f, 0.8125f,
+        0.833333f, 0.854167f,  0.875f,     0.895833f, 0.916667f,
+        0.9375f,   0.958333f,  0.979167f  // Band 18
+    }};
+
+}  // namespace
+
+SpectralCorrelator::SpectralCorrelator()
+    : weights_(kOpusBandWeights24kHz20ms.begin(),
+               kOpusBandWeights24kHz20ms.end()) {}
+
+SpectralCorrelator::~SpectralCorrelator() = default;
+
+void SpectralCorrelator::ComputeAutoCorrelation(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kOpusBands24kHz> auto_corr) const {
+  ComputeCrossCorrelation(x, x, auto_corr);
+}
+
+void SpectralCorrelator::ComputeCrossCorrelation(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<const float> y,
+    rtc::ArrayView<float, kOpusBands24kHz> cross_corr) const {
+  RTC_DCHECK_EQ(x.size(), kFrameSize20ms24kHz);
+  RTC_DCHECK_EQ(x.size(), y.size());
+  RTC_DCHECK_EQ(x[1], 0.f) << "The Nyquist coefficient must be zeroed.";
+  RTC_DCHECK_EQ(y[1], 0.f) << "The Nyquist coefficient must be zeroed.";
+  constexpr auto kOpusScaleNumBins24kHz20ms = GetOpusScaleNumBins24kHz20ms();
+  int k = 0;  // Next Fourier coefficient index.
+  cross_corr[0] = 0.f;
+  for (int i = 0; i < kOpusBands24kHz - 1; ++i) {
+    cross_corr[i + 1] = 0.f;
+    for (int j = 0; j < kOpusScaleNumBins24kHz20ms[i]; ++j) {  // Band size.
+      const float v = x[2 * k] * y[2 * k] + x[2 * k + 1] * y[2 * k + 1];
+      const float tmp = weights_[k] * v;
+      cross_corr[i] += v - tmp;
+      cross_corr[i + 1] += tmp;
+      k++;
+    }
+  }
+  cross_corr[0] *= 2.f;  // The first band only gets half contribution.
+  RTC_DCHECK_EQ(k, kFrameSize20ms24kHz / 2);  // Nyquist coefficient never used.
+}
+
+void ComputeSmoothedLogMagnitudeSpectrum(
+    rtc::ArrayView<const float> bands_energy,
+    rtc::ArrayView<float, kNumBands> log_bands_energy) {
+  RTC_DCHECK_LE(bands_energy.size(), kNumBands);
+  constexpr float kOneByHundred = 1e-2f;
+  constexpr float kLogOneByHundred = -2.f;
+  // Init.
+  float log_max = kLogOneByHundred;
+  float follow = kLogOneByHundred;
+  const auto smooth = [&log_max, &follow](float x) {
+    x = std::max(log_max - 7.f, std::max(follow - 1.5f, x));
+    log_max = std::max(log_max, x);
+    follow = std::max(follow - 1.5f, x);
+    return x;
+  };
+  // Smoothing over the bands for which the band energy is defined.
+  for (int i = 0; rtc::SafeLt(i, bands_energy.size()); ++i) {
+    log_bands_energy[i] = smooth(std::log10(kOneByHundred + bands_energy[i]));
+  }
+  // Smoothing over the remaining bands (zero energy).
+  for (int i = bands_energy.size(); i < kNumBands; ++i) {
+    log_bands_energy[i] = smooth(kLogOneByHundred);
+  }
+}
+
+std::array<float, kNumBands * kNumBands> ComputeDctTable() {
+  std::array<float, kNumBands * kNumBands> dct_table;
+  const double k = std::sqrt(0.5);
+  for (int i = 0; i < kNumBands; ++i) {
+    for (int j = 0; j < kNumBands; ++j)
+      dct_table[i * kNumBands + j] = std::cos((i + 0.5) * j * kPi / kNumBands);
+    dct_table[i * kNumBands] *= k;
+  }
+  return dct_table;
+}
+
+void ComputeDct(rtc::ArrayView<const float> in,
+                rtc::ArrayView<const float, kNumBands * kNumBands> dct_table,
+                rtc::ArrayView<float> out) {
+  // DCT scaling factor - i.e., sqrt(2 / kNumBands).
+  constexpr float kDctScalingFactor = 0.301511345f;
+  constexpr float kDctScalingFactorError =
+      kDctScalingFactor * kDctScalingFactor -
+      2.f / static_cast<float>(kNumBands);
+  static_assert(
+      (kDctScalingFactorError >= 0.f && kDctScalingFactorError < 1e-1f) ||
+          (kDctScalingFactorError < 0.f && kDctScalingFactorError > -1e-1f),
+      "kNumBands changed and kDctScalingFactor has not been updated.");
+  RTC_DCHECK_NE(in.data(), out.data()) << "In-place DCT is not supported.";
+  RTC_DCHECK_LE(in.size(), kNumBands);
+  RTC_DCHECK_LE(1, out.size());
+  RTC_DCHECK_LE(out.size(), in.size());
+  for (int i = 0; rtc::SafeLt(i, out.size()); ++i) {
+    out[i] = 0.f;
+    for (int j = 0; rtc::SafeLt(j, in.size()); ++j) {
+      out[i] += in[j] * dct_table[j * kNumBands + i];
+    }
+    // TODO(bugs.webrtc.org/10480): Scaling factor in the DCT table.
+    out[i] *= kDctScalingFactor;
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
new file mode 100644
index 0000000000..f4b293a567
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
@@ -0,0 +1,100 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// At a sample rate of 24 kHz, the last 3 Opus bands are beyond the Nyquist
+// frequency. However, band #19 gets the contributions from band #18 because
+// of the symmetric triangular filter with peak response at 12 kHz.
+constexpr int kOpusBands24kHz = 20;
+static_assert(kOpusBands24kHz < kNumBands,
+              "The number of bands at 24 kHz must be less than those defined "
+              "in the Opus scale at 48 kHz.");
+
+// Number of FFT frequency bins covered by each band in the Opus scale at a
+// sample rate of 24 kHz for 20 ms frames.
+// Declared here for unit testing.
+constexpr std::array<int, kOpusBands24kHz - 1> GetOpusScaleNumBins24kHz20ms() {
+  return {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 16, 16, 16, 24, 24, 32, 48};
+}
+
+// TODO(bugs.webrtc.org/10480): Move to a separate file.
+// Class to compute band-wise spectral features in the Opus perceptual scale
+// for 20 ms frames sampled at 24 kHz. The analysis methods apply triangular
+// filters with peak response at the each band boundary.
+class SpectralCorrelator {
+ public:
+  // Ctor.
+  SpectralCorrelator();
+  SpectralCorrelator(const SpectralCorrelator&) = delete;
+  SpectralCorrelator& operator=(const SpectralCorrelator&) = delete;
+  ~SpectralCorrelator();
+
+  // Computes the band-wise spectral auto-correlations.
+  // `x` must:
+  //  - have size equal to `kFrameSize20ms24kHz`;
+  //  - be encoded as vectors of interleaved real-complex FFT coefficients
+  //    where x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
+  void ComputeAutoCorrelation(
+      rtc::ArrayView<const float> x,
+      rtc::ArrayView<float, kOpusBands24kHz> auto_corr) const;
+
+  // Computes the band-wise spectral cross-correlations.
+  // `x` and `y` must:
+  //  - have size equal to `kFrameSize20ms24kHz`;
+  //  - be encoded as vectors of interleaved real-complex FFT coefficients where
+  //    x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
+  void ComputeCrossCorrelation(
+      rtc::ArrayView<const float> x,
+      rtc::ArrayView<const float> y,
+      rtc::ArrayView<float, kOpusBands24kHz> cross_corr) const;
+
+ private:
+  const std::vector<float> weights_;  // Weights for each Fourier coefficient.
+};
+
+// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
+// spectral_features.cc. Given a vector of Opus-bands energy coefficients,
+// computes the log magnitude spectrum applying smoothing both over time and
+// over frequency. Declared here for unit testing.
+void ComputeSmoothedLogMagnitudeSpectrum(
+    rtc::ArrayView<const float> bands_energy,
+    rtc::ArrayView<float, kNumBands> log_bands_energy);
+
+// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
+// spectral_features.cc. Creates a DCT table for arrays having size equal to
+// `kNumBands`. Declared here for unit testing.
+std::array<float, kNumBands * kNumBands> ComputeDctTable();
+
+// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
+// spectral_features.cc. Computes DCT for `in` given a pre-computed DCT table.
+// In-place computation is not allowed and `out` can be smaller than `in` in
+// order to only compute the first DCT coefficients. Declared here for unit
+// testing.
+void ComputeDct(rtc::ArrayView<const float> in,
+                rtc::ArrayView<const float, kNumBands * kNumBands> dct_table,
+                rtc::ArrayView<float> out);
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
new file mode 100644
index 0000000000..ece4eb5024
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  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/rnn_vad/spectral_features_internal.h"
+
+#include <algorithm>
+#include <array>
+#include <complex>
+#include <numeric>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+#include "rtc_base/numerics/safe_compare.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Generates the values for the array named `kOpusBandWeights24kHz20ms` in the
+// anonymous namespace of the .cc file, which is the array of FFT coefficient
+// weights for the Opus scale triangular filters.
+std::vector<float> ComputeTriangularFiltersWeights() {
+  constexpr auto kOpusScaleNumBins24kHz20ms = GetOpusScaleNumBins24kHz20ms();
+  const auto& v = kOpusScaleNumBins24kHz20ms;  // Alias.
+  const int num_weights = std::accumulate(kOpusScaleNumBins24kHz20ms.begin(),
+                                          kOpusScaleNumBins24kHz20ms.end(), 0);
+  std::vector<float> weights(num_weights);
+  int next_fft_coeff_index = 0;
+  for (int band = 0; rtc::SafeLt(band, v.size()); ++band) {
+    const int band_size = v[band];
+    for (int j = 0; rtc::SafeLt(j, band_size); ++j) {
+      weights[next_fft_coeff_index + j] = static_cast<float>(j) / band_size;
+    }
+    next_fft_coeff_index += band_size;
+  }
+  return weights;
+}
+
+// Checks that the values returned by GetOpusScaleNumBins24kHz20ms() match the
+// Opus scale frequency boundaries.
+TEST(RnnVadTest, TestOpusScaleBoundaries) {
+  constexpr int kBandFrequencyBoundariesHz[kNumBands - 1] = {
+      200,  400,  600,  800,  1000, 1200, 1400, 1600,  2000,  2400, 2800,
+      3200, 4000, 4800, 5600, 6800, 8000, 9600, 12000, 15600, 20000};
+  constexpr auto kOpusScaleNumBins24kHz20ms = GetOpusScaleNumBins24kHz20ms();
+  int prev = 0;
+  for (int i = 0; rtc::SafeLt(i, kOpusScaleNumBins24kHz20ms.size()); ++i) {
+    int boundary =
+        kBandFrequencyBoundariesHz[i] * kFrameSize20ms24kHz / kSampleRate24kHz;
+    EXPECT_EQ(kOpusScaleNumBins24kHz20ms[i], boundary - prev);
+    prev = boundary;
+  }
+}
+
+// Checks that the computed triangular filters weights for the Opus scale are
+// monotonic withing each Opus band. This test should only be enabled when
+// ComputeTriangularFiltersWeights() is changed and `kOpusBandWeights24kHz20ms`
+// is updated accordingly.
+TEST(RnnVadTest, DISABLED_TestOpusScaleWeights) {
+  auto weights = ComputeTriangularFiltersWeights();
+  int i = 0;
+  for (int band_size : GetOpusScaleNumBins24kHz20ms()) {
+    SCOPED_TRACE(band_size);
+    rtc::ArrayView<float> band_weights(weights.data() + i, band_size);
+    float prev = -1.f;
+    for (float weight : band_weights) {
+      EXPECT_LT(prev, weight);
+      prev = weight;
+    }
+    i += band_size;
+  }
+}
+
+// Checks that the computed band-wise auto-correlation is non-negative for a
+// simple input vector of FFT coefficients.
+TEST(RnnVadTest, SpectralCorrelatorValidOutput) {
+  // Input: vector of (1, 1j) values.
+  Pffft fft(kFrameSize20ms24kHz, Pffft::FftType::kReal);
+  auto in = fft.CreateBuffer();
+  std::array<float, kOpusBands24kHz> out;
+  auto in_view = in->GetView();
+  std::fill(in_view.begin(), in_view.end(), 1.f);
+  in_view[1] = 0.f;  // Nyquist frequency.
+  // Compute and check output.
+  SpectralCorrelator e;
+  e.ComputeAutoCorrelation(in_view, out);
+  for (int i = 0; i < kOpusBands24kHz; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_GT(out[i], 0.f);
+  }
+}
+
+// Checks that the computed smoothed log magnitude spectrum is within tolerance
+// given hard-coded test input data.
+TEST(RnnVadTest, ComputeSmoothedLogMagnitudeSpectrumWithinTolerance) {
+  constexpr std::array<float, kNumBands> input = {
+      {86.060539245605f, 275.668334960938f, 43.406528472900f, 6.541896820068f,
+       17.964015960693f, 8.090919494629f,   1.261920094490f,  1.212702631950f,
+       1.619154453278f,  0.508935272694f,   0.346316039562f,  0.237035423517f,
+       0.172424271703f,  0.271657168865f,   0.126088857651f,  0.139967113733f,
+       0.207200810313f,  0.155893072486f,   0.091090843081f,  0.033391401172f,
+       0.013879744336f,  0.011973354965f}};
+  constexpr std::array<float, kNumBands> expected_output = {
+      {1.934854507446f,  2.440402746201f,  1.637655138969f,  0.816367030144f,
+       1.254645109177f,  0.908534288406f,  0.104459829628f,  0.087320849299f,
+       0.211962252855f,  -0.284886807203f, -0.448164641857f, -0.607240796089f,
+       -0.738917350769f, -0.550279200077f, -0.866177439690f, -0.824003994465f,
+       -0.663138568401f, -0.780171751976f, -0.995288193226f, -1.362596273422f,
+       -1.621970295906f, -1.658103585243f}};
+  std::array<float, kNumBands> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    ComputeSmoothedLogMagnitudeSpectrum(input, computed_output);
+    ExpectNearAbsolute(expected_output, computed_output, 1e-5f);
+  }
+}
+
+// Checks that the computed DCT is within tolerance given hard-coded test input
+// data.
+TEST(RnnVadTest, ComputeDctWithinTolerance) {
+  constexpr std::array<float, kNumBands> input = {
+      {0.232155621052f,  0.678957760334f, 0.220818966627f,  -0.077363930643f,
+       -0.559227049351f, 0.432545185089f, 0.353900641203f,  0.398993015289f,
+       0.409774333239f,  0.454977899790f, 0.300520688295f,  -0.010286616161f,
+       0.272525429726f,  0.098067551851f, 0.083649002016f,  0.046226885170f,
+       -0.033228103071f, 0.144773483276f, -0.117661058903f, -0.005628800020f,
+       -0.009547689930f, -0.045382082462f}};
+  constexpr std::array<float, kNumBands> expected_output = {
+      {0.697072803974f,  0.442710995674f,  -0.293156713247f, -0.060711503029f,
+       0.292050391436f,  0.489301353693f,  0.402255415916f,  0.134404733777f,
+       -0.086305990815f, -0.199605688453f, -0.234511867166f, -0.413774639368f,
+       -0.388507157564f, -0.032798115164f, 0.044605545700f,  0.112466648221f,
+       -0.050096966326f, 0.045971218497f,  -0.029815061018f, -0.410366982222f,
+       -0.209233760834f, -0.128037497401f}};
+  auto dct_table = ComputeDctTable();
+  std::array<float, kNumBands> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    ComputeDct(input, dct_table, computed_output);
+    ExpectNearAbsolute(expected_output, computed_output, 1e-5f);
+  }
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
new file mode 100644
index 0000000000..324d694957
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  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/rnn_vad/spectral_features.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kTestFeatureVectorSize = kNumBands + 3 * kNumLowerBands + 1;
+
+// Writes non-zero sample values.
+void WriteTestData(rtc::ArrayView<float> samples) {
+  for (int i = 0; rtc::SafeLt(i, samples.size()); ++i) {
+    samples[i] = i % 100;
+  }
+}
+
+rtc::ArrayView<float, kNumBands - kNumLowerBands> GetHigherBandsSpectrum(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumLowerBands, kNumBands - kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetAverage(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data(), kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetFirstDerivative(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumBands, kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetSecondDerivative(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumBands + kNumLowerBands, kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetCepstralCrossCorrelation(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumBands + 2 * kNumLowerBands,
+          kNumLowerBands};
+}
+
+float* GetCepstralVariability(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return feature_vector->data() + kNumBands + 3 * kNumLowerBands;
+}
+
+constexpr float kInitialFeatureVal = -9999.f;
+
+// Checks that silence is detected when the input signal is 0 and that the
+// feature vector is written only if the input signal is not tagged as silence.
+TEST(RnnVadTest, SpectralFeaturesWithAndWithoutSilence) {
+  // Initialize.
+  SpectralFeaturesExtractor sfe;
+  std::array<float, kFrameSize20ms24kHz> samples;
+  rtc::ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
+  bool is_silence;
+  std::array<float, kTestFeatureVectorSize> feature_vector;
+
+  // Write an initial value in the feature vector to detect changes.
+  std::fill(feature_vector.begin(), feature_vector.end(), kInitialFeatureVal);
+
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+
+  // With silence.
+  std::fill(samples.begin(), samples.end(), 0.f);
+  is_silence = sfe.CheckSilenceComputeFeatures(
+      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
+      GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
+      GetSecondDerivative(&feature_vector),
+      GetCepstralCrossCorrelation(&feature_vector),
+      GetCepstralVariability(&feature_vector));
+  // Silence is expected, the output won't be overwritten.
+  EXPECT_TRUE(is_silence);
+  EXPECT_TRUE(std::all_of(feature_vector.begin(), feature_vector.end(),
+                          [](float x) { return x == kInitialFeatureVal; }));
+
+  // With no silence.
+  WriteTestData(samples);
+  is_silence = sfe.CheckSilenceComputeFeatures(
+      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
+      GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
+      GetSecondDerivative(&feature_vector),
+      GetCepstralCrossCorrelation(&feature_vector),
+      GetCepstralVariability(&feature_vector));
+  // Silence is not expected, the output will be overwritten.
+  EXPECT_FALSE(is_silence);
+  EXPECT_FALSE(std::all_of(feature_vector.begin(), feature_vector.end(),
+                           [](float x) { return x == kInitialFeatureVal; }));
+}
+
+// Feeds a constant input signal and checks that:
+// - the cepstral coefficients average does not change;
+// - the derivatives are zero;
+// - the cepstral variability score does not change.
+TEST(RnnVadTest, CepstralFeaturesConstantAverageZeroDerivative) {
+  // Initialize.
+  SpectralFeaturesExtractor sfe;
+  std::array<float, kFrameSize20ms24kHz> samples;
+  rtc::ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
+  WriteTestData(samples);
+
+  // Fill the spectral features with test data.
+  std::array<float, kTestFeatureVectorSize> feature_vector;
+  for (int i = 0; i < kCepstralCoeffsHistorySize; ++i) {
+    sfe.CheckSilenceComputeFeatures(
+        samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
+        GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
+        GetSecondDerivative(&feature_vector),
+        GetCepstralCrossCorrelation(&feature_vector),
+        GetCepstralVariability(&feature_vector));
+  }
+
+  // Feed the test data one last time but using a different output vector.
+  std::array<float, kTestFeatureVectorSize> feature_vector_last;
+  sfe.CheckSilenceComputeFeatures(
+      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector_last),
+      GetAverage(&feature_vector_last),
+      GetFirstDerivative(&feature_vector_last),
+      GetSecondDerivative(&feature_vector_last),
+      GetCepstralCrossCorrelation(&feature_vector_last),
+      GetCepstralVariability(&feature_vector_last));
+
+  // Average is unchanged.
+  ExpectEqualFloatArray({feature_vector.data(), kNumLowerBands},
+                        {feature_vector_last.data(), kNumLowerBands});
+  // First and second derivatives are zero.
+  constexpr std::array<float, kNumLowerBands> zeros{};
+  ExpectEqualFloatArray(
+      {feature_vector_last.data() + kNumBands, kNumLowerBands}, zeros);
+  ExpectEqualFloatArray(
+      {feature_vector_last.data() + kNumBands + kNumLowerBands, kNumLowerBands},
+      zeros);
+  // Variability is unchanged.
+  EXPECT_FLOAT_EQ(feature_vector[kNumBands + 3 * kNumLowerBands],
+                  feature_vector_last[kNumBands + 3 * kNumLowerBands]);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
new file mode 100644
index 0000000000..d186479551
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
@@ -0,0 +1,95 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
+
+#include <algorithm>
+#include <array>
+#include <cstring>
+#include <utility>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Data structure to buffer the results of pair-wise comparisons between items
+// stored in a ring buffer. Every time that the oldest item is replaced in the
+// ring buffer, the new one is compared to the remaining items in the ring
+// buffer. The results of such comparisons need to be buffered and automatically
+// removed when one of the two corresponding items that have been compared is
+// removed from the ring buffer. It is assumed that the comparison is symmetric
+// and that comparing an item with itself is not needed.
+template <typename T, int S>
+class SymmetricMatrixBuffer {
+  static_assert(S > 2, "");
+
+ public:
+  SymmetricMatrixBuffer() = default;
+  SymmetricMatrixBuffer(const SymmetricMatrixBuffer&) = delete;
+  SymmetricMatrixBuffer& operator=(const SymmetricMatrixBuffer&) = delete;
+  ~SymmetricMatrixBuffer() = default;
+  // Sets the buffer values to zero.
+  void Reset() {
+    static_assert(std::is_arithmetic<T>::value,
+                  "Integral or floating point required.");
+    buf_.fill(0);
+  }
+  // Pushes the results from the comparison between the most recent item and
+  // those that are still in the ring buffer. The first element in `values` must
+  // correspond to the comparison between the most recent item and the second
+  // most recent one in the ring buffer, whereas the last element in `values`
+  // must correspond to the comparison between the most recent item and the
+  // oldest one in the ring buffer.
+  void Push(rtc::ArrayView<T, S - 1> values) {
+    // Move the lower-right sub-matrix of size (S-2) x (S-2) one row up and one
+    // column left.
+    std::memmove(buf_.data(), buf_.data() + S, (buf_.size() - S) * sizeof(T));
+    // Copy new values in the last column in the right order.
+    for (int i = 0; rtc::SafeLt(i, values.size()); ++i) {
+      const int index = (S - 1 - i) * (S - 1) - 1;
+      RTC_DCHECK_GE(index, 0);
+      RTC_DCHECK_LT(index, buf_.size());
+      buf_[index] = values[i];
+    }
+  }
+  // Reads the value that corresponds to comparison of two items in the ring
+  // buffer having delay `delay1` and `delay2`. The two arguments must not be
+  // equal and both must be in {0, ..., S - 1}.
+  T GetValue(int delay1, int delay2) const {
+    int row = S - 1 - delay1;
+    int col = S - 1 - delay2;
+    RTC_DCHECK_NE(row, col) << "The diagonal cannot be accessed.";
+    if (row > col)
+      std::swap(row, col);  // Swap to access the upper-right triangular part.
+    RTC_DCHECK_LE(0, row);
+    RTC_DCHECK_LT(row, S - 1) << "Not enforcing row < col and row != col.";
+    RTC_DCHECK_LE(1, col) << "Not enforcing row < col and row != col.";
+    RTC_DCHECK_LT(col, S);
+    const int index = row * (S - 1) + (col - 1);
+    RTC_DCHECK_LE(0, index);
+    RTC_DCHECK_LT(index, buf_.size());
+    return buf_[index];
+  }
+
+ private:
+  // Encode an upper-right triangular matrix (excluding its diagonal) using a
+  // square matrix. This allows to move the data in Push() with one single
+  // operation.
+  std::array<T, (S - 1) * (S - 1)> buf_{};
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
new file mode 100644
index 0000000000..1509ca5ac1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
@@ -0,0 +1,107 @@
+/*
+ *  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/rnn_vad/symmetric_matrix_buffer.h"
+
+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+template <typename T, int S>
+void CheckSymmetry(const SymmetricMatrixBuffer<T, S>* sym_matrix_buf) {
+  for (int row = 0; row < S - 1; ++row)
+    for (int col = row + 1; col < S; ++col)
+      EXPECT_EQ(sym_matrix_buf->GetValue(row, col),
+                sym_matrix_buf->GetValue(col, row));
+}
+
+using PairType = std::pair<int, int>;
+
+// Checks that the symmetric matrix buffer contains any pair with a value equal
+// to the given one.
+template <int S>
+bool CheckPairsWithValueExist(
+    const SymmetricMatrixBuffer<PairType, S>* sym_matrix_buf,
+    const int value) {
+  for (int row = 0; row < S - 1; ++row) {
+    for (int col = row + 1; col < S; ++col) {
+      auto p = sym_matrix_buf->GetValue(row, col);
+      if (p.first == value || p.second == value)
+        return true;
+    }
+  }
+  return false;
+}
+
+// Test that shows how to combine RingBuffer and SymmetricMatrixBuffer to
+// efficiently compute pair-wise scores. This test verifies that the evolution
+// of a SymmetricMatrixBuffer instance follows that of RingBuffer.
+TEST(RnnVadTest, SymmetricMatrixBufferUseCase) {
+  // Instance a ring buffer which will be fed with a series of integer values.
+  constexpr int kRingBufSize = 10;
+  RingBuffer<int, 1, kRingBufSize> ring_buf;
+  // Instance a symmetric matrix buffer for the ring buffer above. It stores
+  // pairs of integers with which this test can easily check that the evolution
+  // of RingBuffer and SymmetricMatrixBuffer match.
+  SymmetricMatrixBuffer<PairType, kRingBufSize> sym_matrix_buf;
+  for (int t = 1; t <= 100; ++t) {  // Evolution steps.
+    SCOPED_TRACE(t);
+    const int t_removed = ring_buf.GetArrayView(kRingBufSize - 1)[0];
+    ring_buf.Push({&t, 1});
+    // The head of the ring buffer is `t`.
+    ASSERT_EQ(t, ring_buf.GetArrayView(0)[0]);
+    // Create the comparisons between `t` and the older elements in the ring
+    // buffer.
+    std::array<PairType, kRingBufSize - 1> new_comparions;
+    for (int i = 0; i < kRingBufSize - 1; ++i) {
+      // Start comparing `t` to the second newest element in the ring buffer.
+      const int delay = i + 1;
+      const auto t_prev = ring_buf.GetArrayView(delay)[0];
+      ASSERT_EQ(std::max(0, t - delay), t_prev);
+      // Compare the last element `t` with `t_prev`.
+      new_comparions[i].first = t_prev;
+      new_comparions[i].second = t;
+    }
+    // Push the new comparisons in the symmetric matrix buffer.
+    sym_matrix_buf.Push({new_comparions.data(), new_comparions.size()});
+    // Tests.
+    CheckSymmetry(&sym_matrix_buf);
+    // Check that the pairs resulting from the content in the ring buffer are
+    // in the right position.
+    for (int delay1 = 0; delay1 < kRingBufSize - 1; ++delay1) {
+      for (int delay2 = delay1 + 1; delay2 < kRingBufSize; ++delay2) {
+        const auto t1 = ring_buf.GetArrayView(delay1)[0];
+        const auto t2 = ring_buf.GetArrayView(delay2)[0];
+        ASSERT_LE(t2, t1);
+        const auto p = sym_matrix_buf.GetValue(delay1, delay2);
+        EXPECT_EQ(p.first, t2);
+        EXPECT_EQ(p.second, t1);
+      }
+    }
+    // Check that every older element in the ring buffer still has a
+    // corresponding pair in the symmetric matrix buffer.
+    for (int delay = 1; delay < kRingBufSize; ++delay) {
+      const auto t_prev = ring_buf.GetArrayView(delay)[0];
+      EXPECT_TRUE(CheckPairsWithValueExist(&sym_matrix_buf, t_prev));
+    }
+    // Check that the element removed from the ring buffer has no corresponding
+    // pairs in the symmetric matrix buffer.
+    if (t > kRingBufSize - 1) {
+      EXPECT_FALSE(CheckPairsWithValueExist(&sym_matrix_buf, t_removed));
+    }
+  }
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc
new file mode 100644
index 0000000000..857a9f2706
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc
@@ -0,0 +1,143 @@
+/*
+ *  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/rnn_vad/test_utils.h"
+
+#include <algorithm>
+#include <fstream>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// File reader for binary files that contain a sequence of values with
+// arithmetic type `T`. The values of type `T` that are read are cast to float.
+template <typename T>
+class FloatFileReader : public FileReader {
+ public:
+  static_assert(std::is_arithmetic<T>::value, "");
+  explicit FloatFileReader(absl::string_view filename)
+      : is_(std::string(filename), std::ios::binary | std::ios::ate),
+        size_(is_.tellg() / sizeof(T)) {
+    RTC_CHECK(is_);
+    SeekBeginning();
+  }
+  FloatFileReader(const FloatFileReader&) = delete;
+  FloatFileReader& operator=(const FloatFileReader&) = delete;
+  ~FloatFileReader() = default;
+
+  int size() const override { return size_; }
+  bool ReadChunk(rtc::ArrayView<float> dst) override {
+    const std::streamsize bytes_to_read = dst.size() * sizeof(T);
+    if (std::is_same<T, float>::value) {
+      is_.read(reinterpret_cast<char*>(dst.data()), bytes_to_read);
+    } else {
+      buffer_.resize(dst.size());
+      is_.read(reinterpret_cast<char*>(buffer_.data()), bytes_to_read);
+      std::transform(buffer_.begin(), buffer_.end(), dst.begin(),
+                     [](const T& v) -> float { return static_cast<float>(v); });
+    }
+    return is_.gcount() == bytes_to_read;
+  }
+  bool ReadValue(float& dst) override { return ReadChunk({&dst, 1}); }
+  void SeekForward(int hop) override { is_.seekg(hop * sizeof(T), is_.cur); }
+  void SeekBeginning() override { is_.seekg(0, is_.beg); }
+
+ private:
+  std::ifstream is_;
+  const int size_;
+  std::vector<T> buffer_;
+};
+
+}  // namespace
+
+using webrtc::test::ResourcePath;
+
+void ExpectEqualFloatArray(rtc::ArrayView<const float> expected,
+                           rtc::ArrayView<const float> computed) {
+  ASSERT_EQ(expected.size(), computed.size());
+  for (int i = 0; rtc::SafeLt(i, expected.size()); ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_FLOAT_EQ(expected[i], computed[i]);
+  }
+}
+
+void ExpectNearAbsolute(rtc::ArrayView<const float> expected,
+                        rtc::ArrayView<const float> computed,
+                        float tolerance) {
+  ASSERT_EQ(expected.size(), computed.size());
+  for (int i = 0; rtc::SafeLt(i, expected.size()); ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_NEAR(expected[i], computed[i], tolerance);
+  }
+}
+
+std::unique_ptr<FileReader> CreatePcmSamplesReader() {
+  return std::make_unique<FloatFileReader<int16_t>>(
+      /*filename=*/test::ResourcePath("audio_processing/agc2/rnn_vad/samples",
+                                      "pcm"));
+}
+
+ChunksFileReader CreatePitchBuffer24kHzReader() {
+  auto reader = std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath(
+          "audio_processing/agc2/rnn_vad/pitch_buf_24k", "dat"));
+  const int num_chunks = rtc::CheckedDivExact(reader->size(), kBufSize24kHz);
+  return {/*chunk_size=*/kBufSize24kHz, num_chunks, std::move(reader)};
+}
+
+ChunksFileReader CreateLpResidualAndPitchInfoReader() {
+  constexpr int kPitchInfoSize = 2;  // Pitch period and strength.
+  constexpr int kChunkSize = kBufSize24kHz + kPitchInfoSize;
+  auto reader = std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath(
+          "audio_processing/agc2/rnn_vad/pitch_lp_res", "dat"));
+  const int num_chunks = rtc::CheckedDivExact(reader->size(), kChunkSize);
+  return {kChunkSize, num_chunks, std::move(reader)};
+}
+
+std::unique_ptr<FileReader> CreateGruInputReader() {
+  return std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath("audio_processing/agc2/rnn_vad/gru_in",
+                                      "dat"));
+}
+
+std::unique_ptr<FileReader> CreateVadProbsReader() {
+  return std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath("audio_processing/agc2/rnn_vad/vad_prob",
+                                      "dat"));
+}
+
+PitchTestData::PitchTestData() {
+  FloatFileReader<float> reader(
+      /*filename=*/ResourcePath(
+          "audio_processing/agc2/rnn_vad/pitch_search_int", "dat"));
+  reader.ReadChunk(pitch_buffer_24k_);
+  reader.ReadChunk(square_energies_24k_);
+  reader.ReadChunk(auto_correlation_12k_);
+  // Reverse the order of the squared energy values.
+  // Required after the WebRTC CL 191703 which switched to forward computation.
+  std::reverse(square_energies_24k_.begin(), square_energies_24k_.end());
+}
+
+PitchTestData::~PitchTestData() = default;
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h
new file mode 100644
index 0000000000..e64b7b7ecd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h
@@ -0,0 +1,130 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
+
+#include <array>
+#include <fstream>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+constexpr float kFloatMin = std::numeric_limits<float>::min();
+
+// Fails for every pair from two equally sized rtc::ArrayView<float> views such
+// that the values in the pair do not match.
+void ExpectEqualFloatArray(rtc::ArrayView<const float> expected,
+                           rtc::ArrayView<const float> computed);
+
+// Fails for every pair from two equally sized rtc::ArrayView<float> views such
+// that their absolute error is above a given threshold.
+void ExpectNearAbsolute(rtc::ArrayView<const float> expected,
+                        rtc::ArrayView<const float> computed,
+                        float tolerance);
+
+// File reader interface.
+class FileReader {
+ public:
+  virtual ~FileReader() = default;
+  // Number of values in the file.
+  virtual int size() const = 0;
+  // Reads `dst.size()` float values into `dst`, advances the internal file
+  // position according to the number of read bytes and returns true if the
+  // values are correctly read. If the number of remaining bytes in the file is
+  // not sufficient to read `dst.size()` float values, `dst` is partially
+  // modified and false is returned.
+  virtual bool ReadChunk(rtc::ArrayView<float> dst) = 0;
+  // Reads a single float value, advances the internal file position according
+  // to the number of read bytes and returns true if the value is correctly
+  // read. If the number of remaining bytes in the file is not sufficient to
+  // read one float, `dst` is not modified and false is returned.
+  virtual bool ReadValue(float& dst) = 0;
+  // Advances the internal file position by `hop` float values.
+  virtual void SeekForward(int hop) = 0;
+  // Resets the internal file position to BOF.
+  virtual void SeekBeginning() = 0;
+};
+
+// File reader for files that contain `num_chunks` chunks with size equal to
+// `chunk_size`.
+struct ChunksFileReader {
+  const int chunk_size;
+  const int num_chunks;
+  std::unique_ptr<FileReader> reader;
+};
+
+// Creates a reader for the PCM S16 samples file.
+std::unique_ptr<FileReader> CreatePcmSamplesReader();
+
+// Creates a reader for the 24 kHz pitch buffer test data.
+ChunksFileReader CreatePitchBuffer24kHzReader();
+
+// Creates a reader for the LP residual and pitch information test data.
+ChunksFileReader CreateLpResidualAndPitchInfoReader();
+
+// Creates a reader for the sequence of GRU input vectors.
+std::unique_ptr<FileReader> CreateGruInputReader();
+
+// Creates a reader for the VAD probabilities test data.
+std::unique_ptr<FileReader> CreateVadProbsReader();
+
+// Class to retrieve a test pitch buffer content and the expected output for the
+// analysis steps.
+class PitchTestData {
+ public:
+  PitchTestData();
+  ~PitchTestData();
+  rtc::ArrayView<const float, kBufSize24kHz> PitchBuffer24kHzView() const {
+    return pitch_buffer_24k_;
+  }
+  rtc::ArrayView<const float, kRefineNumLags24kHz> SquareEnergies24kHzView()
+      const {
+    return square_energies_24k_;
+  }
+  rtc::ArrayView<const float, kNumLags12kHz> AutoCorrelation12kHzView() const {
+    return auto_correlation_12k_;
+  }
+
+ private:
+  std::array<float, kBufSize24kHz> pitch_buffer_24k_;
+  std::array<float, kRefineNumLags24kHz> square_energies_24k_;
+  std::array<float, kNumLags12kHz> auto_correlation_12k_;
+};
+
+// Writer for binary files.
+class FileWriter {
+ public:
+  explicit FileWriter(absl::string_view file_path)
+      : os_(std::string(file_path), std::ios::binary) {}
+  FileWriter(const FileWriter&) = delete;
+  FileWriter& operator=(const FileWriter&) = delete;
+  ~FileWriter() = default;
+  void WriteChunk(rtc::ArrayView<const float> value) {
+    const std::streamsize bytes_to_write = value.size() * sizeof(float);
+    os_.write(reinterpret_cast<const char*>(value.data()), bytes_to_write);
+  }
+
+ private:
+  std::ofstream os_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math.h
new file mode 100644
index 0000000000..47f681196a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_VECTOR_MATH_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_VECTOR_MATH_H_
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_HAS_NEON)
+#include <arm_neon.h>
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+
+#include <numeric>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Provides optimizations for mathematical operations having vectors as
+// operand(s).
+class VectorMath {
+ public:
+  explicit VectorMath(AvailableCpuFeatures cpu_features)
+      : cpu_features_(cpu_features) {}
+
+  // Computes the dot product between two equally sized vectors.
+  float DotProduct(rtc::ArrayView<const float> x,
+                   rtc::ArrayView<const float> y) const {
+    RTC_DCHECK_EQ(x.size(), y.size());
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    if (cpu_features_.avx2) {
+      return DotProductAvx2(x, y);
+    } else if (cpu_features_.sse2) {
+      __m128 accumulator = _mm_setzero_ps();
+      constexpr int kBlockSizeLog2 = 2;
+      constexpr int kBlockSize = 1 << kBlockSizeLog2;
+      const int incomplete_block_index = (x.size() >> kBlockSizeLog2)
+                                         << kBlockSizeLog2;
+      for (int i = 0; i < incomplete_block_index; i += kBlockSize) {
+        RTC_DCHECK_LE(i + kBlockSize, x.size());
+        const __m128 x_i = _mm_loadu_ps(&x[i]);
+        const __m128 y_i = _mm_loadu_ps(&y[i]);
+        // Multiply-add.
+        const __m128 z_j = _mm_mul_ps(x_i, y_i);
+        accumulator = _mm_add_ps(accumulator, z_j);
+      }
+      // Reduce `accumulator` by addition.
+      __m128 high = _mm_movehl_ps(accumulator, accumulator);
+      accumulator = _mm_add_ps(accumulator, high);
+      high = _mm_shuffle_ps(accumulator, accumulator, 1);
+      accumulator = _mm_add_ps(accumulator, high);
+      float dot_product = _mm_cvtss_f32(accumulator);
+      // Add the result for the last block if incomplete.
+      for (int i = incomplete_block_index;
+           i < rtc::dchecked_cast<int>(x.size()); ++i) {
+        dot_product += x[i] * y[i];
+      }
+      return dot_product;
+    }
+#elif defined(WEBRTC_HAS_NEON) && defined(WEBRTC_ARCH_ARM64)
+    if (cpu_features_.neon) {
+      float32x4_t accumulator = vdupq_n_f32(0.f);
+      constexpr int kBlockSizeLog2 = 2;
+      constexpr int kBlockSize = 1 << kBlockSizeLog2;
+      const int incomplete_block_index = (x.size() >> kBlockSizeLog2)
+                                         << kBlockSizeLog2;
+      for (int i = 0; i < incomplete_block_index; i += kBlockSize) {
+        RTC_DCHECK_LE(i + kBlockSize, x.size());
+        const float32x4_t x_i = vld1q_f32(&x[i]);
+        const float32x4_t y_i = vld1q_f32(&y[i]);
+        accumulator = vfmaq_f32(accumulator, x_i, y_i);
+      }
+      // Reduce `accumulator` by addition.
+      const float32x2_t tmp =
+          vpadd_f32(vget_low_f32(accumulator), vget_high_f32(accumulator));
+      float dot_product = vget_lane_f32(vpadd_f32(tmp, vrev64_f32(tmp)), 0);
+      // Add the result for the last block if incomplete.
+      for (int i = incomplete_block_index;
+           i < rtc::dchecked_cast<int>(x.size()); ++i) {
+        dot_product += x[i] * y[i];
+      }
+      return dot_product;
+    }
+#endif
+    return std::inner_product(x.begin(), x.end(), y.begin(), 0.f);
+  }
+
+ private:
+  float DotProductAvx2(rtc::ArrayView<const float> x,
+                       rtc::ArrayView<const float> y) const;
+
+  const AvailableCpuFeatures cpu_features_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_VECTOR_MATH_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc
new file mode 100644
index 0000000000..a875e11daf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2020 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 <immintrin.h>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+float VectorMath::DotProductAvx2(rtc::ArrayView<const float> x,
+                                 rtc::ArrayView<const float> y) const {
+  RTC_DCHECK(cpu_features_.avx2);
+  RTC_DCHECK_EQ(x.size(), y.size());
+  __m256 accumulator = _mm256_setzero_ps();
+  constexpr int kBlockSizeLog2 = 3;
+  constexpr int kBlockSize = 1 << kBlockSizeLog2;
+  const int incomplete_block_index = (x.size() >> kBlockSizeLog2)
+                                     << kBlockSizeLog2;
+  for (int i = 0; i < incomplete_block_index; i += kBlockSize) {
+    RTC_DCHECK_LE(i + kBlockSize, x.size());
+    const __m256 x_i = _mm256_loadu_ps(&x[i]);
+    const __m256 y_i = _mm256_loadu_ps(&y[i]);
+    accumulator = _mm256_fmadd_ps(x_i, y_i, accumulator);
+  }
+  // Reduce `accumulator` by addition.
+  __m128 high = _mm256_extractf128_ps(accumulator, 1);
+  __m128 low = _mm256_extractf128_ps(accumulator, 0);
+  low = _mm_add_ps(high, low);
+  high = _mm_movehl_ps(high, low);
+  low = _mm_add_ps(high, low);
+  high = _mm_shuffle_ps(low, low, 1);
+  low = _mm_add_ss(high, low);
+  float dot_product = _mm_cvtss_f32(low);
+  // Add the result for the last block if incomplete.
+  for (int i = incomplete_block_index; i < rtc::dchecked_cast<int>(x.size());
+       ++i) {
+    dot_product += x[i] * y[i];
+  }
+  return dot_product;
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build
new file mode 100644
index 0000000000..d4dd169f15
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build
@@ -0,0 +1,189 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+CXXFLAGS += [
+    "-mavx2",
+    "-mfma"
+]
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_AVX2"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+Library("vector_math_avx2_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build
new file mode 100644
index 0000000000..09fe0c3d24
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build
@@ -0,0 +1,220 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("vector_math_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_unittest.cc
new file mode 100644
index 0000000000..45fd65d61e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_unittest.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2020 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/rnn_vad/vector_math.h"
+
+#include <vector>
+
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kSizeOfX = 19;
+constexpr float kX[kSizeOfX] = {
+    0.31593041f, 0.9350786f,   -0.25252445f, -0.86956251f, -0.9673632f,
+    0.54571901f, -0.72504495f, -0.79509912f, -0.25525012f, -0.73340473f,
+    0.15747377f, -0.04370565f, 0.76135145f,  -0.57239645f, 0.68616848f,
+    0.3740298f,  0.34710799f,  -0.92207423f, 0.10738454f};
+constexpr int kSizeOfXSubSpan = 16;
+static_assert(kSizeOfXSubSpan < kSizeOfX, "");
+constexpr float kEnergyOfX = 7.315563958160327f;
+constexpr float kEnergyOfXSubspan = 6.333327669592963f;
+
+class VectorMathParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+TEST_P(VectorMathParametrization, TestDotProduct) {
+  VectorMath vector_math(/*cpu_features=*/GetParam());
+  EXPECT_FLOAT_EQ(vector_math.DotProduct(kX, kX), kEnergyOfX);
+  EXPECT_FLOAT_EQ(
+      vector_math.DotProduct({kX, kSizeOfXSubSpan}, {kX, kSizeOfXSubSpan}),
+      kEnergyOfXSubspan);
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/false});
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    VectorMathParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc