Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

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

19 files changed, 2393 insertions, 0 deletions

diff --git a/third_party/libwebrtc/common_audio/vad/include/vad.h b/third_party/libwebrtc/common_audio/vad/include/vad.h
new file mode 100644
index 0000000000..b15275b166
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/include/vad.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2014 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 COMMON_AUDIO_VAD_INCLUDE_VAD_H_
+#define COMMON_AUDIO_VAD_INCLUDE_VAD_H_
+
+#include <memory>
+
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class Vad {
+ public:
+  enum Aggressiveness {
+    kVadNormal = 0,
+    kVadLowBitrate = 1,
+    kVadAggressive = 2,
+    kVadVeryAggressive = 3
+  };
+
+  enum Activity { kPassive = 0, kActive = 1, kError = -1 };
+
+  virtual ~Vad() = default;
+
+  // Calculates a VAD decision for the given audio frame. Valid sample rates
+  // are 8000, 16000, and 32000 Hz; the number of samples must be such that the
+  // frame is 10, 20, or 30 ms long.
+  virtual Activity VoiceActivity(const int16_t* audio,
+                                 size_t num_samples,
+                                 int sample_rate_hz) = 0;
+
+  // Resets VAD state.
+  virtual void Reset() = 0;
+};
+
+// Returns a Vad instance that's implemented on top of WebRtcVad.
+std::unique_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness);
+
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_VAD_INCLUDE_VAD_H_
diff --git a/third_party/libwebrtc/common_audio/vad/include/webrtc_vad.h b/third_party/libwebrtc/common_audio/vad/include/webrtc_vad.h
new file mode 100644
index 0000000000..31e628f058
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/include/webrtc_vad.h
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+/*
+ * This header file includes the VAD API calls. Specific function calls are
+ * given below.
+ */
+
+#ifndef COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
+#define COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+typedef struct WebRtcVadInst VadInst;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Creates an instance to the VAD structure.
+VadInst* WebRtcVad_Create(void);
+
+// Frees the dynamic memory of a specified VAD instance.
+//
+// - handle [i] : Pointer to VAD instance that should be freed.
+void WebRtcVad_Free(VadInst* handle);
+
+// Initializes a VAD instance.
+//
+// - handle [i/o] : Instance that should be initialized.
+//
+// returns        : 0 - (OK),
+//                 -1 - (null pointer or Default mode could not be set).
+int WebRtcVad_Init(VadInst* handle);
+
+// Sets the VAD operating mode. A more aggressive (higher mode) VAD is more
+// restrictive in reporting speech. Put in other words the probability of being
+// speech when the VAD returns 1 is increased with increasing mode. As a
+// consequence also the missed detection rate goes up.
+//
+// - handle [i/o] : VAD instance.
+// - mode   [i]   : Aggressiveness mode (0, 1, 2, or 3).
+//
+// returns        : 0 - (OK),
+//                 -1 - (null pointer, mode could not be set or the VAD instance
+//                       has not been initialized).
+int WebRtcVad_set_mode(VadInst* handle, int mode);
+
+// Calculates a VAD decision for the `audio_frame`. For valid sampling rates
+// frame lengths, see the description of WebRtcVad_ValidRatesAndFrameLengths().
+//
+// - handle       [i/o] : VAD Instance. Needs to be initialized by
+//                        WebRtcVad_Init() before call.
+// - fs           [i]   : Sampling frequency (Hz): 8000, 16000, or 32000
+// - audio_frame  [i]   : Audio frame buffer.
+// - frame_length [i]   : Length of audio frame buffer in number of samples.
+//
+// returns              : 1 - (Active Voice),
+//                        0 - (Non-active Voice),
+//                       -1 - (Error)
+int WebRtcVad_Process(VadInst* handle,
+                      int fs,
+                      const int16_t* audio_frame,
+                      size_t frame_length);
+
+// Checks for valid combinations of `rate` and `frame_length`. We support 10,
+// 20 and 30 ms frames and the rates 8000, 16000 and 32000 Hz.
+//
+// - rate         [i] : Sampling frequency (Hz).
+// - frame_length [i] : Speech frame buffer length in number of samples.
+//
+// returns            : 0 - (valid combination), -1 - (invalid combination)
+int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
diff --git a/third_party/libwebrtc/common_audio/vad/mock/mock_vad.h b/third_party/libwebrtc/common_audio/vad/mock/mock_vad.h
new file mode 100644
index 0000000000..5a554ce1f9
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/mock/mock_vad.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2014 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 COMMON_AUDIO_VAD_MOCK_MOCK_VAD_H_
+#define COMMON_AUDIO_VAD_MOCK_MOCK_VAD_H_
+
+#include "common_audio/vad/include/vad.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockVad : public Vad {
+ public:
+  ~MockVad() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+
+  MOCK_METHOD(enum Activity,
+              VoiceActivity,
+              (const int16_t* audio, size_t num_samples, int sample_rate_hz),
+              (override));
+  MOCK_METHOD(void, Reset, (), (override));
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_VAD_MOCK_MOCK_VAD_H_
diff --git a/third_party/libwebrtc/common_audio/vad/vad.cc b/third_party/libwebrtc/common_audio/vad/vad.cc
new file mode 100644
index 0000000000..1647246590
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2014 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 "common_audio/vad/include/vad.h"
+
+#include <memory>
+
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+class VadImpl final : public Vad {
+ public:
+  explicit VadImpl(Aggressiveness aggressiveness)
+      : handle_(nullptr), aggressiveness_(aggressiveness) {
+    Reset();
+  }
+
+  ~VadImpl() override { WebRtcVad_Free(handle_); }
+
+  Activity VoiceActivity(const int16_t* audio,
+                         size_t num_samples,
+                         int sample_rate_hz) override {
+    int ret = WebRtcVad_Process(handle_, sample_rate_hz, audio, num_samples);
+    switch (ret) {
+      case 0:
+        return kPassive;
+      case 1:
+        return kActive;
+      default:
+        RTC_DCHECK_NOTREACHED() << "WebRtcVad_Process returned an error.";
+        return kError;
+    }
+  }
+
+  void Reset() override {
+    if (handle_)
+      WebRtcVad_Free(handle_);
+    handle_ = WebRtcVad_Create();
+    RTC_CHECK(handle_);
+    RTC_CHECK_EQ(WebRtcVad_Init(handle_), 0);
+    RTC_CHECK_EQ(WebRtcVad_set_mode(handle_, aggressiveness_), 0);
+  }
+
+ private:
+  VadInst* handle_;
+  Aggressiveness aggressiveness_;
+};
+
+}  // namespace
+
+std::unique_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness) {
+  return std::unique_ptr<Vad>(new VadImpl(aggressiveness));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/common_audio/vad/vad_core.c b/third_party/libwebrtc/common_audio/vad/vad_core.c
new file mode 100644
index 0000000000..0872449a7c
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_core.c
@@ -0,0 +1,685 @@
+/*
+ *  Copyright (c) 2012 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 "common_audio/vad/vad_core.h"
+
+#include "rtc_base/sanitizer.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/vad/vad_filterbank.h"
+#include "common_audio/vad/vad_gmm.h"
+#include "common_audio/vad/vad_sp.h"
+
+// Spectrum Weighting
+static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };
+static const int16_t kNoiseUpdateConst = 655; // Q15
+static const int16_t kSpeechUpdateConst = 6554; // Q15
+static const int16_t kBackEta = 154; // Q8
+// Minimum difference between the two models, Q5
+static const int16_t kMinimumDifference[kNumChannels] = {
+    544, 544, 576, 576, 576, 576 };
+// Upper limit of mean value for speech model, Q7
+static const int16_t kMaximumSpeech[kNumChannels] = {
+    11392, 11392, 11520, 11520, 11520, 11520 };
+// Minimum value for mean value
+static const int16_t kMinimumMean[kNumGaussians] = { 640, 768 };
+// Upper limit of mean value for noise model, Q7
+static const int16_t kMaximumNoise[kNumChannels] = {
+    9216, 9088, 8960, 8832, 8704, 8576 };
+// Start values for the Gaussian models, Q7
+// Weights for the two Gaussians for the six channels (noise)
+static const int16_t kNoiseDataWeights[kTableSize] = {
+    34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
+// Weights for the two Gaussians for the six channels (speech)
+static const int16_t kSpeechDataWeights[kTableSize] = {
+    48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
+// Means for the two Gaussians for the six channels (noise)
+static const int16_t kNoiseDataMeans[kTableSize] = {
+    6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
+// Means for the two Gaussians for the six channels (speech)
+static const int16_t kSpeechDataMeans[kTableSize] = {
+    8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
+};
+// Stds for the two Gaussians for the six channels (noise)
+static const int16_t kNoiseDataStds[kTableSize] = {
+    378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
+// Stds for the two Gaussians for the six channels (speech)
+static const int16_t kSpeechDataStds[kTableSize] = {
+    555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
+
+// Constants used in GmmProbability().
+//
+// Maximum number of counted speech (VAD = 1) frames in a row.
+static const int16_t kMaxSpeechFrames = 6;
+// Minimum standard deviation for both speech and noise.
+static const int16_t kMinStd = 384;
+
+// Constants in WebRtcVad_InitCore().
+// Default aggressiveness mode.
+static const short kDefaultMode = 0;
+static const int kInitCheck = 42;
+
+// Constants used in WebRtcVad_set_mode_core().
+//
+// Thresholds for different frame lengths (10 ms, 20 ms and 30 ms).
+//
+// Mode 0, Quality.
+static const int16_t kOverHangMax1Q[3] = { 8, 4, 3 };
+static const int16_t kOverHangMax2Q[3] = { 14, 7, 5 };
+static const int16_t kLocalThresholdQ[3] = { 24, 21, 24 };
+static const int16_t kGlobalThresholdQ[3] = { 57, 48, 57 };
+// Mode 1, Low bitrate.
+static const int16_t kOverHangMax1LBR[3] = { 8, 4, 3 };
+static const int16_t kOverHangMax2LBR[3] = { 14, 7, 5 };
+static const int16_t kLocalThresholdLBR[3] = { 37, 32, 37 };
+static const int16_t kGlobalThresholdLBR[3] = { 100, 80, 100 };
+// Mode 2, Aggressive.
+static const int16_t kOverHangMax1AGG[3] = { 6, 3, 2 };
+static const int16_t kOverHangMax2AGG[3] = { 9, 5, 3 };
+static const int16_t kLocalThresholdAGG[3] = { 82, 78, 82 };
+static const int16_t kGlobalThresholdAGG[3] = { 285, 260, 285 };
+// Mode 3, Very aggressive.
+static const int16_t kOverHangMax1VAG[3] = { 6, 3, 2 };
+static const int16_t kOverHangMax2VAG[3] = { 9, 5, 3 };
+static const int16_t kLocalThresholdVAG[3] = { 94, 94, 94 };
+static const int16_t kGlobalThresholdVAG[3] = { 1100, 1050, 1100 };
+
+// Calculates the weighted average w.r.t. number of Gaussians. The `data` are
+// updated with an `offset` before averaging.
+//
+// - data     [i/o] : Data to average.
+// - offset   [i]   : An offset added to `data`.
+// - weights  [i]   : Weights used for averaging.
+//
+// returns          : The weighted average.
+static int32_t WeightedAverage(int16_t* data, int16_t offset,
+                               const int16_t* weights) {
+  int k;
+  int32_t weighted_average = 0;
+
+  for (k = 0; k < kNumGaussians; k++) {
+    data[k * kNumChannels] += offset;
+    weighted_average += data[k * kNumChannels] * weights[k * kNumChannels];
+  }
+  return weighted_average;
+}
+
+// An s16 x s32 -> s32 multiplication that's allowed to overflow. (It's still
+// undefined behavior, so not a good idea; this just makes UBSan ignore the
+// violation, so that our old code can continue to do what it's always been
+// doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingMulS16ByS32ToS32(int16_t a, int32_t b) {
+  return a * b;
+}
+
+// Calculates the probabilities for both speech and background noise using
+// Gaussian Mixture Models (GMM). A hypothesis-test is performed to decide which
+// type of signal is most probable.
+//
+// - self           [i/o] : Pointer to VAD instance
+// - features       [i]   : Feature vector of length `kNumChannels`
+//                          = log10(energy in frequency band)
+// - total_power    [i]   : Total power in audio frame.
+// - frame_length   [i]   : Number of input samples
+//
+// - returns              : the VAD decision (0 - noise, 1 - speech).
+static int16_t GmmProbability(VadInstT* self, int16_t* features,
+                              int16_t total_power, size_t frame_length) {
+  int channel, k;
+  int16_t feature_minimum;
+  int16_t h0, h1;
+  int16_t log_likelihood_ratio;
+  int16_t vadflag = 0;
+  int16_t shifts_h0, shifts_h1;
+  int16_t tmp_s16, tmp1_s16, tmp2_s16;
+  int16_t diff;
+  int gaussian;
+  int16_t nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
+  int16_t delt, ndelt;
+  int16_t maxspe, maxmu;
+  int16_t deltaN[kTableSize], deltaS[kTableSize];
+  int16_t ngprvec[kTableSize] = { 0 };  // Conditional probability = 0.
+  int16_t sgprvec[kTableSize] = { 0 };  // Conditional probability = 0.
+  int32_t h0_test, h1_test;
+  int32_t tmp1_s32, tmp2_s32;
+  int32_t sum_log_likelihood_ratios = 0;
+  int32_t noise_global_mean, speech_global_mean;
+  int32_t noise_probability[kNumGaussians], speech_probability[kNumGaussians];
+  int16_t overhead1, overhead2, individualTest, totalTest;
+
+  // Set various thresholds based on frame lengths (80, 160 or 240 samples).
+  if (frame_length == 80) {
+    overhead1 = self->over_hang_max_1[0];
+    overhead2 = self->over_hang_max_2[0];
+    individualTest = self->individual[0];
+    totalTest = self->total[0];
+  } else if (frame_length == 160) {
+    overhead1 = self->over_hang_max_1[1];
+    overhead2 = self->over_hang_max_2[1];
+    individualTest = self->individual[1];
+    totalTest = self->total[1];
+  } else {
+    overhead1 = self->over_hang_max_1[2];
+    overhead2 = self->over_hang_max_2[2];
+    individualTest = self->individual[2];
+    totalTest = self->total[2];
+  }
+
+  if (total_power > kMinEnergy) {
+    // The signal power of current frame is large enough for processing. The
+    // processing consists of two parts:
+    // 1) Calculating the likelihood of speech and thereby a VAD decision.
+    // 2) Updating the underlying model, w.r.t., the decision made.
+
+    // The detection scheme is an LRT with hypothesis
+    // H0: Noise
+    // H1: Speech
+    //
+    // We combine a global LRT with local tests, for each frequency sub-band,
+    // here defined as `channel`.
+    for (channel = 0; channel < kNumChannels; channel++) {
+      // For each channel we model the probability with a GMM consisting of
+      // `kNumGaussians`, with different means and standard deviations depending
+      // on H0 or H1.
+      h0_test = 0;
+      h1_test = 0;
+      for (k = 0; k < kNumGaussians; k++) {
+        gaussian = channel + k * kNumChannels;
+        // Probability under H0, that is, probability of frame being noise.
+        // Value given in Q27 = Q7 * Q20.
+        tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
+                                                 self->noise_means[gaussian],
+                                                 self->noise_stds[gaussian],
+                                                 &deltaN[gaussian]);
+        noise_probability[k] = kNoiseDataWeights[gaussian] * tmp1_s32;
+        h0_test += noise_probability[k];  // Q27
+
+        // Probability under H1, that is, probability of frame being speech.
+        // Value given in Q27 = Q7 * Q20.
+        tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
+                                                 self->speech_means[gaussian],
+                                                 self->speech_stds[gaussian],
+                                                 &deltaS[gaussian]);
+        speech_probability[k] = kSpeechDataWeights[gaussian] * tmp1_s32;
+        h1_test += speech_probability[k];  // Q27
+      }
+
+      // Calculate the log likelihood ratio: log2(Pr{X|H1} / Pr{X|H1}).
+      // Approximation:
+      // log2(Pr{X|H1} / Pr{X|H1}) = log2(Pr{X|H1}*2^Q) - log2(Pr{X|H1}*2^Q)
+      //                           = log2(h1_test) - log2(h0_test)
+      //                           = log2(2^(31-shifts_h1)*(1+b1))
+      //                             - log2(2^(31-shifts_h0)*(1+b0))
+      //                           = shifts_h0 - shifts_h1
+      //                             + log2(1+b1) - log2(1+b0)
+      //                          ~= shifts_h0 - shifts_h1
+      //
+      // Note that b0 and b1 are values less than 1, hence, 0 <= log2(1+b0) < 1.
+      // Further, b0 and b1 are independent and on the average the two terms
+      // cancel.
+      shifts_h0 = WebRtcSpl_NormW32(h0_test);
+      shifts_h1 = WebRtcSpl_NormW32(h1_test);
+      if (h0_test == 0) {
+        shifts_h0 = 31;
+      }
+      if (h1_test == 0) {
+        shifts_h1 = 31;
+      }
+      log_likelihood_ratio = shifts_h0 - shifts_h1;
+
+      // Update `sum_log_likelihood_ratios` with spectrum weighting. This is
+      // used for the global VAD decision.
+      sum_log_likelihood_ratios +=
+          (int32_t) (log_likelihood_ratio * kSpectrumWeight[channel]);
+
+      // Local VAD decision.
+      if ((log_likelihood_ratio * 4) > individualTest) {
+        vadflag = 1;
+      }
+
+      // TODO(bjornv): The conditional probabilities below are applied on the
+      // hard coded number of Gaussians set to two. Find a way to generalize.
+      // Calculate local noise probabilities used later when updating the GMM.
+      h0 = (int16_t) (h0_test >> 12);  // Q15
+      if (h0 > 0) {
+        // High probability of noise. Assign conditional probabilities for each
+        // Gaussian in the GMM.
+        tmp1_s32 = (noise_probability[0] & 0xFFFFF000) << 2;  // Q29
+        ngprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h0);  // Q14
+        ngprvec[channel + kNumChannels] = 16384 - ngprvec[channel];
+      } else {
+        // Low noise probability. Assign conditional probability 1 to the first
+        // Gaussian and 0 to the rest (which is already set at initialization).
+        ngprvec[channel] = 16384;
+      }
+
+      // Calculate local speech probabilities used later when updating the GMM.
+      h1 = (int16_t) (h1_test >> 12);  // Q15
+      if (h1 > 0) {
+        // High probability of speech. Assign conditional probabilities for each
+        // Gaussian in the GMM. Otherwise use the initialized values, i.e., 0.
+        tmp1_s32 = (speech_probability[0] & 0xFFFFF000) << 2;  // Q29
+        sgprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h1);  // Q14
+        sgprvec[channel + kNumChannels] = 16384 - sgprvec[channel];
+      }
+    }
+
+    // Make a global VAD decision.
+    vadflag |= (sum_log_likelihood_ratios >= totalTest);
+
+    // Update the model parameters.
+    maxspe = 12800;
+    for (channel = 0; channel < kNumChannels; channel++) {
+
+      // Get minimum value in past which is used for long term correction in Q4.
+      feature_minimum = WebRtcVad_FindMinimum(self, features[channel], channel);
+
+      // Compute the "global" mean, that is the sum of the two means weighted.
+      noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
+                                          &kNoiseDataWeights[channel]);
+      tmp1_s16 = (int16_t) (noise_global_mean >> 6);  // Q8
+
+      for (k = 0; k < kNumGaussians; k++) {
+        gaussian = channel + k * kNumChannels;
+
+        nmk = self->noise_means[gaussian];
+        smk = self->speech_means[gaussian];
+        nsk = self->noise_stds[gaussian];
+        ssk = self->speech_stds[gaussian];
+
+        // Update noise mean vector if the frame consists of noise only.
+        nmk2 = nmk;
+        if (!vadflag) {
+          // deltaN = (x-mu)/sigma^2
+          // ngprvec[k] = `noise_probability[k]` /
+          //   (`noise_probability[0]` + `noise_probability[1]`)
+
+          // (Q14 * Q11 >> 11) = Q14.
+          delt = (int16_t)((ngprvec[gaussian] * deltaN[gaussian]) >> 11);
+          // Q7 + (Q14 * Q15 >> 22) = Q7.
+          nmk2 = nmk + (int16_t)((delt * kNoiseUpdateConst) >> 22);
+        }
+
+        // Long term correction of the noise mean.
+        // Q8 - Q8 = Q8.
+        ndelt = (feature_minimum << 4) - tmp1_s16;
+        // Q7 + (Q8 * Q8) >> 9 = Q7.
+        nmk3 = nmk2 + (int16_t)((ndelt * kBackEta) >> 9);
+
+        // Control that the noise mean does not drift to much.
+        tmp_s16 = (int16_t) ((k + 5) << 7);
+        if (nmk3 < tmp_s16) {
+          nmk3 = tmp_s16;
+        }
+        tmp_s16 = (int16_t) ((72 + k - channel) << 7);
+        if (nmk3 > tmp_s16) {
+          nmk3 = tmp_s16;
+        }
+        self->noise_means[gaussian] = nmk3;
+
+        if (vadflag) {
+          // Update speech mean vector:
+          // `deltaS` = (x-mu)/sigma^2
+          // sgprvec[k] = `speech_probability[k]` /
+          //   (`speech_probability[0]` + `speech_probability[1]`)
+
+          // (Q14 * Q11) >> 11 = Q14.
+          delt = (int16_t)((sgprvec[gaussian] * deltaS[gaussian]) >> 11);
+          // Q14 * Q15 >> 21 = Q8.
+          tmp_s16 = (int16_t)((delt * kSpeechUpdateConst) >> 21);
+          // Q7 + (Q8 >> 1) = Q7. With rounding.
+          smk2 = smk + ((tmp_s16 + 1) >> 1);
+
+          // Control that the speech mean does not drift to much.
+          maxmu = maxspe + 640;
+          if (smk2 < kMinimumMean[k]) {
+            smk2 = kMinimumMean[k];
+          }
+          if (smk2 > maxmu) {
+            smk2 = maxmu;
+          }
+          self->speech_means[gaussian] = smk2;  // Q7.
+
+          // (Q7 >> 3) = Q4. With rounding.
+          tmp_s16 = ((smk + 4) >> 3);
+
+          tmp_s16 = features[channel] - tmp_s16;  // Q4
+          // (Q11 * Q4 >> 3) = Q12.
+          tmp1_s32 = (deltaS[gaussian] * tmp_s16) >> 3;
+          tmp2_s32 = tmp1_s32 - 4096;
+          tmp_s16 = sgprvec[gaussian] >> 2;
+          // (Q14 >> 2) * Q12 = Q24.
+          tmp1_s32 = tmp_s16 * tmp2_s32;
+
+          tmp2_s32 = tmp1_s32 >> 4;  // Q20
+
+          // 0.1 * Q20 / Q7 = Q13.
+          if (tmp2_s32 > 0) {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp2_s32, ssk * 10);
+          } else {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp2_s32, ssk * 10);
+            tmp_s16 = -tmp_s16;
+          }
+          // Divide by 4 giving an update factor of 0.025 (= 0.1 / 4).
+          // Note that division by 4 equals shift by 2, hence,
+          // (Q13 >> 8) = (Q13 >> 6) / 4 = Q7.
+          tmp_s16 += 128;  // Rounding.
+          ssk += (tmp_s16 >> 8);
+          if (ssk < kMinStd) {
+            ssk = kMinStd;
+          }
+          self->speech_stds[gaussian] = ssk;
+        } else {
+          // Update GMM variance vectors.
+          // deltaN * (features[channel] - nmk) - 1
+          // Q4 - (Q7 >> 3) = Q4.
+          tmp_s16 = features[channel] - (nmk >> 3);
+          // (Q11 * Q4 >> 3) = Q12.
+          tmp1_s32 = (deltaN[gaussian] * tmp_s16) >> 3;
+          tmp1_s32 -= 4096;
+
+          // (Q14 >> 2) * Q12 = Q24.
+          tmp_s16 = (ngprvec[gaussian] + 2) >> 2;
+          tmp2_s32 = OverflowingMulS16ByS32ToS32(tmp_s16, tmp1_s32);
+          // Q20  * approx 0.001 (2^-10=0.0009766), hence,
+          // (Q24 >> 14) = (Q24 >> 4) / 2^10 = Q20.
+          tmp1_s32 = tmp2_s32 >> 14;
+
+          // Q20 / Q7 = Q13.
+          if (tmp1_s32 > 0) {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, nsk);
+          } else {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp1_s32, nsk);
+            tmp_s16 = -tmp_s16;
+          }
+          tmp_s16 += 32;  // Rounding
+          nsk += tmp_s16 >> 6;  // Q13 >> 6 = Q7.
+          if (nsk < kMinStd) {
+            nsk = kMinStd;
+          }
+          self->noise_stds[gaussian] = nsk;
+        }
+      }
+
+      // Separate models if they are too close.
+      // `noise_global_mean` in Q14 (= Q7 * Q7).
+      noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
+                                          &kNoiseDataWeights[channel]);
+
+      // `speech_global_mean` in Q14 (= Q7 * Q7).
+      speech_global_mean = WeightedAverage(&self->speech_means[channel], 0,
+                                           &kSpeechDataWeights[channel]);
+
+      // `diff` = "global" speech mean - "global" noise mean.
+      // (Q14 >> 9) - (Q14 >> 9) = Q5.
+      diff = (int16_t) (speech_global_mean >> 9) -
+          (int16_t) (noise_global_mean >> 9);
+      if (diff < kMinimumDifference[channel]) {
+        tmp_s16 = kMinimumDifference[channel] - diff;
+
+        // `tmp1_s16` = ~0.8 * (kMinimumDifference - diff) in Q7.
+        // `tmp2_s16` = ~0.2 * (kMinimumDifference - diff) in Q7.
+        tmp1_s16 = (int16_t)((13 * tmp_s16) >> 2);
+        tmp2_s16 = (int16_t)((3 * tmp_s16) >> 2);
+
+        // Move Gaussian means for speech model by `tmp1_s16` and update
+        // `speech_global_mean`. Note that `self->speech_means[channel]` is
+        // changed after the call.
+        speech_global_mean = WeightedAverage(&self->speech_means[channel],
+                                             tmp1_s16,
+                                             &kSpeechDataWeights[channel]);
+
+        // Move Gaussian means for noise model by -`tmp2_s16` and update
+        // `noise_global_mean`. Note that `self->noise_means[channel]` is
+        // changed after the call.
+        noise_global_mean = WeightedAverage(&self->noise_means[channel],
+                                            -tmp2_s16,
+                                            &kNoiseDataWeights[channel]);
+      }
+
+      // Control that the speech & noise means do not drift to much.
+      maxspe = kMaximumSpeech[channel];
+      tmp2_s16 = (int16_t) (speech_global_mean >> 7);
+      if (tmp2_s16 > maxspe) {
+        // Upper limit of speech model.
+        tmp2_s16 -= maxspe;
+
+        for (k = 0; k < kNumGaussians; k++) {
+          self->speech_means[channel + k * kNumChannels] -= tmp2_s16;
+        }
+      }
+
+      tmp2_s16 = (int16_t) (noise_global_mean >> 7);
+      if (tmp2_s16 > kMaximumNoise[channel]) {
+        tmp2_s16 -= kMaximumNoise[channel];
+
+        for (k = 0; k < kNumGaussians; k++) {
+          self->noise_means[channel + k * kNumChannels] -= tmp2_s16;
+        }
+      }
+    }
+    self->frame_counter++;
+  }
+
+  // Smooth with respect to transition hysteresis.
+  if (!vadflag) {
+    if (self->over_hang > 0) {
+      vadflag = 2 + self->over_hang;
+      self->over_hang--;
+    }
+    self->num_of_speech = 0;
+  } else {
+    self->num_of_speech++;
+    if (self->num_of_speech > kMaxSpeechFrames) {
+      self->num_of_speech = kMaxSpeechFrames;
+      self->over_hang = overhead2;
+    } else {
+      self->over_hang = overhead1;
+    }
+  }
+  return vadflag;
+}
+
+// Initialize the VAD. Set aggressiveness mode to default value.
+int WebRtcVad_InitCore(VadInstT* self) {
+  int i;
+
+  if (self == NULL) {
+    return -1;
+  }
+
+  // Initialization of general struct variables.
+  self->vad = 1;  // Speech active (=1).
+  self->frame_counter = 0;
+  self->over_hang = 0;
+  self->num_of_speech = 0;
+
+  // Initialization of downsampling filter state.
+  memset(self->downsampling_filter_states, 0,
+         sizeof(self->downsampling_filter_states));
+
+  // Initialization of 48 to 8 kHz downsampling.
+  WebRtcSpl_ResetResample48khzTo8khz(&self->state_48_to_8);
+
+  // Read initial PDF parameters.
+  for (i = 0; i < kTableSize; i++) {
+    self->noise_means[i] = kNoiseDataMeans[i];
+    self->speech_means[i] = kSpeechDataMeans[i];
+    self->noise_stds[i] = kNoiseDataStds[i];
+    self->speech_stds[i] = kSpeechDataStds[i];
+  }
+
+  // Initialize Index and Minimum value vectors.
+  for (i = 0; i < 16 * kNumChannels; i++) {
+    self->low_value_vector[i] = 10000;
+    self->index_vector[i] = 0;
+  }
+
+  // Initialize splitting filter states.
+  memset(self->upper_state, 0, sizeof(self->upper_state));
+  memset(self->lower_state, 0, sizeof(self->lower_state));
+
+  // Initialize high pass filter states.
+  memset(self->hp_filter_state, 0, sizeof(self->hp_filter_state));
+
+  // Initialize mean value memory, for WebRtcVad_FindMinimum().
+  for (i = 0; i < kNumChannels; i++) {
+    self->mean_value[i] = 1600;
+  }
+
+  // Set aggressiveness mode to default (=`kDefaultMode`).
+  if (WebRtcVad_set_mode_core(self, kDefaultMode) != 0) {
+    return -1;
+  }
+
+  self->init_flag = kInitCheck;
+
+  return 0;
+}
+
+// Set aggressiveness mode
+int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
+  int return_value = 0;
+
+  switch (mode) {
+    case 0:
+      // Quality mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1Q,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2Q,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdQ,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdQ,
+             sizeof(self->total));
+      break;
+    case 1:
+      // Low bitrate mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1LBR,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2LBR,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdLBR,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdLBR,
+             sizeof(self->total));
+      break;
+    case 2:
+      // Aggressive mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1AGG,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2AGG,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdAGG,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdAGG,
+             sizeof(self->total));
+      break;
+    case 3:
+      // Very aggressive mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1VAG,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2VAG,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdVAG,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdVAG,
+             sizeof(self->total));
+      break;
+    default:
+      return_value = -1;
+      break;
+  }
+
+  return return_value;
+}
+
+// Calculate VAD decision by first extracting feature values and then calculate
+// probability for both speech and background noise.
+
+int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
+                           size_t frame_length) {
+  int vad;
+  size_t i;
+  int16_t speech_nb[240];  // 30 ms in 8 kHz.
+  // `tmp_mem` is a temporary memory used by resample function, length is
+  // frame length in 10 ms (480 samples) + 256 extra.
+  int32_t tmp_mem[480 + 256] = { 0 };
+  const size_t kFrameLen10ms48khz = 480;
+  const size_t kFrameLen10ms8khz = 80;
+  size_t num_10ms_frames = frame_length / kFrameLen10ms48khz;
+
+  for (i = 0; i < num_10ms_frames; i++) {
+    WebRtcSpl_Resample48khzTo8khz(speech_frame,
+                                  &speech_nb[i * kFrameLen10ms8khz],
+                                  &inst->state_48_to_8,
+                                  tmp_mem);
+  }
+
+  // Do VAD on an 8 kHz signal
+  vad = WebRtcVad_CalcVad8khz(inst, speech_nb, frame_length / 6);
+
+  return vad;
+}
+
+int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
+                           size_t frame_length)
+{
+    size_t len;
+    int vad;
+    int16_t speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
+    int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
+
+
+    // Downsample signal 32->16->8 before doing VAD
+    WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
+                           frame_length);
+    len = frame_length / 2;
+
+    WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
+    len /= 2;
+
+    // Do VAD on an 8 kHz signal
+    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
+
+    return vad;
+}
+
+int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
+                           size_t frame_length)
+{
+    size_t len;
+    int vad;
+    int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
+
+    // Wideband: Downsample signal before doing VAD
+    WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
+                           frame_length);
+
+    len = frame_length / 2;
+    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
+
+    return vad;
+}
+
+int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
+                          size_t frame_length)
+{
+    int16_t feature_vector[kNumChannels], total_power;
+
+    // Get power in the bands
+    total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
+                                              feature_vector);
+
+    // Make a VAD
+    inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
+
+    return inst->vad;
+}
diff --git a/third_party/libwebrtc/common_audio/vad/vad_core.h b/third_party/libwebrtc/common_audio/vad/vad_core.h
new file mode 100644
index 0000000000..fbaf970065
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_core.h
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+/*
+ * This header file includes the descriptions of the core VAD calls.
+ */
+
+#ifndef COMMON_AUDIO_VAD_VAD_CORE_H_
+#define COMMON_AUDIO_VAD_VAD_CORE_H_
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// TODO(https://bugs.webrtc.org/14476): When converted to C++, remove the macro.
+#if defined(__cplusplus)
+#define CONSTEXPR_INT(x) constexpr int x
+#else
+#define CONSTEXPR_INT(x) enum { x }
+#endif
+
+CONSTEXPR_INT(kNumChannels = 6);  // Number of frequency bands (named channels).
+CONSTEXPR_INT(
+    kNumGaussians = 2);  // Number of Gaussians per channel in the GMM.
+CONSTEXPR_INT(kTableSize = kNumChannels * kNumGaussians);
+CONSTEXPR_INT(
+    kMinEnergy = 10);  // Minimum energy required to trigger audio signal.
+
+typedef struct VadInstT_ {
+  int vad;
+  int32_t downsampling_filter_states[4];
+  WebRtcSpl_State48khzTo8khz state_48_to_8;
+  int16_t noise_means[kTableSize];
+  int16_t speech_means[kTableSize];
+  int16_t noise_stds[kTableSize];
+  int16_t speech_stds[kTableSize];
+  // TODO(bjornv): Change to `frame_count`.
+  int32_t frame_counter;
+  int16_t over_hang;  // Over Hang
+  int16_t num_of_speech;
+  // TODO(bjornv): Change to `age_vector`.
+  int16_t index_vector[16 * kNumChannels];
+  int16_t low_value_vector[16 * kNumChannels];
+  // TODO(bjornv): Change to `median`.
+  int16_t mean_value[kNumChannels];
+  int16_t upper_state[5];
+  int16_t lower_state[5];
+  int16_t hp_filter_state[4];
+  int16_t over_hang_max_1[3];
+  int16_t over_hang_max_2[3];
+  int16_t individual[3];
+  int16_t total[3];
+
+  int init_flag;
+} VadInstT;
+
+// Initializes the core VAD component. The default aggressiveness mode is
+// controlled by `kDefaultMode` in vad_core.c.
+//
+// - self [i/o] : Instance that should be initialized
+//
+// returns      : 0 (OK), -1 (null pointer in or if the default mode can't be
+//                set)
+int WebRtcVad_InitCore(VadInstT* self);
+
+/****************************************************************************
+ * WebRtcVad_set_mode_core(...)
+ *
+ * This function changes the VAD settings
+ *
+ * Input:
+ *      - inst      : VAD instance
+ *      - mode      : Aggressiveness degree
+ *                    0 (High quality) - 3 (Highly aggressive)
+ *
+ * Output:
+ *      - inst      : Changed  instance
+ *
+ * Return value     :  0 - Ok
+ *                    -1 - Error
+ */
+
+int WebRtcVad_set_mode_core(VadInstT* self, int mode);
+
+/****************************************************************************
+ * WebRtcVad_CalcVad48khz(...)
+ * WebRtcVad_CalcVad32khz(...)
+ * WebRtcVad_CalcVad16khz(...)
+ * WebRtcVad_CalcVad8khz(...)
+ *
+ * Calculate probability for active speech and make VAD decision.
+ *
+ * Input:
+ *      - inst          : Instance that should be initialized
+ *      - speech_frame  : Input speech frame
+ *      - frame_length  : Number of input samples
+ *
+ * Output:
+ *      - inst          : Updated filter states etc.
+ *
+ * Return value         : VAD decision
+ *                        0 - No active speech
+ *                        1-6 - Active speech
+ */
+int WebRtcVad_CalcVad48khz(VadInstT* inst,
+                           const int16_t* speech_frame,
+                           size_t frame_length);
+int WebRtcVad_CalcVad32khz(VadInstT* inst,
+                           const int16_t* speech_frame,
+                           size_t frame_length);
+int WebRtcVad_CalcVad16khz(VadInstT* inst,
+                           const int16_t* speech_frame,
+                           size_t frame_length);
+int WebRtcVad_CalcVad8khz(VadInstT* inst,
+                          const int16_t* speech_frame,
+                          size_t frame_length);
+
+#endif  // COMMON_AUDIO_VAD_VAD_CORE_H_
diff --git a/third_party/libwebrtc/common_audio/vad/vad_core_unittest.cc b/third_party/libwebrtc/common_audio/vad/vad_core_unittest.cc
new file mode 100644
index 0000000000..3131a86ae3
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_core_unittest.cc
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2012 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 <stdlib.h>
+
+#include "common_audio/vad/vad_unittest.h"
+#include "test/gtest.h"
+
+extern "C" {
+#include "common_audio/vad/vad_core.h"
+}
+
+namespace webrtc {
+namespace test {
+
+TEST_F(VadTest, InitCore) {
+  // Test WebRtcVad_InitCore().
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+
+  // null pointer test.
+  EXPECT_EQ(-1, WebRtcVad_InitCore(nullptr));
+
+  // Verify return = 0 for non-null pointer.
+  EXPECT_EQ(0, WebRtcVad_InitCore(self));
+  // Verify init_flag is set.
+  EXPECT_EQ(42, self->init_flag);
+
+  free(self);
+}
+
+TEST_F(VadTest, set_mode_core) {
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+
+  // TODO(bjornv): Add null pointer check if we take care of it in
+  // vad_core.c
+
+  ASSERT_EQ(0, WebRtcVad_InitCore(self));
+  // Test WebRtcVad_set_mode_core().
+  // Invalid modes should return -1.
+  EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, -1));
+  EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, 1000));
+  // Valid modes should return 0.
+  for (size_t j = 0; j < kModesSize; ++j) {
+    EXPECT_EQ(0, WebRtcVad_set_mode_core(self, kModes[j]));
+  }
+
+  free(self);
+}
+
+TEST_F(VadTest, CalcVad) {
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+  int16_t speech[kMaxFrameLength];
+
+  // TODO(bjornv): Add null pointer check if we take care of it in
+  // vad_core.c
+
+  // Test WebRtcVad_CalcVadXXkhz()
+  // Verify that all zeros in gives VAD = 0 out.
+  memset(speech, 0, sizeof(speech));
+  ASSERT_EQ(0, WebRtcVad_InitCore(self));
+  for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+    if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(48000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalcVad48khz(self, speech, kFrameLengths[j]));
+    }
+  }
+
+  // Construct a speech signal that will trigger the VAD in all modes. It is
+  // known that (i * i) will wrap around, but that doesn't matter in this case.
+  for (size_t i = 0; i < kMaxFrameLength; ++i) {
+    speech[i] = static_cast<int16_t>(i * i);
+  }
+  for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+    if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+      EXPECT_EQ(1, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
+      EXPECT_EQ(1, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
+      EXPECT_EQ(1, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(48000, kFrameLengths[j])) {
+      EXPECT_EQ(1, WebRtcVad_CalcVad48khz(self, speech, kFrameLengths[j]));
+    }
+  }
+
+  free(self);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/common_audio/vad/vad_filterbank.c b/third_party/libwebrtc/common_audio/vad/vad_filterbank.c
new file mode 100644
index 0000000000..aff63f79cd
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_filterbank.c
@@ -0,0 +1,329 @@
+/*
+ *  Copyright (c) 2012 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 "common_audio/vad/vad_filterbank.h"
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// Constants used in LogOfEnergy().
+static const int16_t kLogConst = 24660;  // 160*log10(2) in Q9.
+static const int16_t kLogEnergyIntPart = 14336;  // 14 in Q10
+
+// Coefficients used by HighPassFilter, Q14.
+static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
+static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
+
+// Allpass filter coefficients, upper and lower, in Q15.
+// Upper: 0.64, Lower: 0.17
+static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
+
+// Adjustment for division with two in SplitFilter.
+static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
+
+// High pass filtering, with a cut-off frequency at 80 Hz, if the `data_in` is
+// sampled at 500 Hz.
+//
+// - data_in      [i]   : Input audio data sampled at 500 Hz.
+// - data_length  [i]   : Length of input and output data.
+// - filter_state [i/o] : State of the filter.
+// - data_out     [o]   : Output audio data in the frequency interval
+//                        80 - 250 Hz.
+static void HighPassFilter(const int16_t* data_in, size_t data_length,
+                           int16_t* filter_state, int16_t* data_out) {
+  size_t i;
+  const int16_t* in_ptr = data_in;
+  int16_t* out_ptr = data_out;
+  int32_t tmp32 = 0;
+
+
+  // The sum of the absolute values of the impulse response:
+  // The zero/pole-filter has a max amplification of a single sample of: 1.4546
+  // Impulse response: 0.4047 -0.6179 -0.0266  0.1993  0.1035  -0.0194
+  // The all-zero section has a max amplification of a single sample of: 1.6189
+  // Impulse response: 0.4047 -0.8094  0.4047  0       0        0
+  // The all-pole section has a max amplification of a single sample of: 1.9931
+  // Impulse response: 1.0000  0.4734 -0.1189 -0.2187 -0.0627   0.04532
+
+  for (i = 0; i < data_length; i++) {
+    // All-zero section (filter coefficients in Q14).
+    tmp32 = kHpZeroCoefs[0] * *in_ptr;
+    tmp32 += kHpZeroCoefs[1] * filter_state[0];
+    tmp32 += kHpZeroCoefs[2] * filter_state[1];
+    filter_state[1] = filter_state[0];
+    filter_state[0] = *in_ptr++;
+
+    // All-pole section (filter coefficients in Q14).
+    tmp32 -= kHpPoleCoefs[1] * filter_state[2];
+    tmp32 -= kHpPoleCoefs[2] * filter_state[3];
+    filter_state[3] = filter_state[2];
+    filter_state[2] = (int16_t) (tmp32 >> 14);
+    *out_ptr++ = filter_state[2];
+  }
+}
+
+// All pass filtering of `data_in`, used before splitting the signal into two
+// frequency bands (low pass vs high pass).
+// Note that `data_in` and `data_out` can NOT correspond to the same address.
+//
+// - data_in            [i]   : Input audio signal given in Q0.
+// - data_length        [i]   : Length of input and output data.
+// - filter_coefficient [i]   : Given in Q15.
+// - filter_state       [i/o] : State of the filter given in Q(-1).
+// - data_out           [o]   : Output audio signal given in Q(-1).
+static void AllPassFilter(const int16_t* data_in, size_t data_length,
+                          int16_t filter_coefficient, int16_t* filter_state,
+                          int16_t* data_out) {
+  // The filter can only cause overflow (in the w16 output variable)
+  // if more than 4 consecutive input numbers are of maximum value and
+  // has the the same sign as the impulse responses first taps.
+  // First 6 taps of the impulse response:
+  // 0.6399 0.5905 -0.3779 0.2418 -0.1547 0.0990
+
+  size_t i;
+  int16_t tmp16 = 0;
+  int32_t tmp32 = 0;
+  int32_t state32 = ((int32_t) (*filter_state) * (1 << 16));  // Q15
+
+  for (i = 0; i < data_length; i++) {
+    tmp32 = state32 + filter_coefficient * *data_in;
+    tmp16 = (int16_t) (tmp32 >> 16);  // Q(-1)
+    *data_out++ = tmp16;
+    state32 = (*data_in * (1 << 14)) - filter_coefficient * tmp16;  // Q14
+    state32 *= 2;  // Q15.
+    data_in += 2;
+  }
+
+  *filter_state = (int16_t) (state32 >> 16);  // Q(-1)
+}
+
+// Splits `data_in` into `hp_data_out` and `lp_data_out` corresponding to
+// an upper (high pass) part and a lower (low pass) part respectively.
+//
+// - data_in      [i]   : Input audio data to be split into two frequency bands.
+// - data_length  [i]   : Length of `data_in`.
+// - upper_state  [i/o] : State of the upper filter, given in Q(-1).
+// - lower_state  [i/o] : State of the lower filter, given in Q(-1).
+// - hp_data_out  [o]   : Output audio data of the upper half of the spectrum.
+//                        The length is `data_length` / 2.
+// - lp_data_out  [o]   : Output audio data of the lower half of the spectrum.
+//                        The length is `data_length` / 2.
+static void SplitFilter(const int16_t* data_in, size_t data_length,
+                        int16_t* upper_state, int16_t* lower_state,
+                        int16_t* hp_data_out, int16_t* lp_data_out) {
+  size_t i;
+  size_t half_length = data_length >> 1;  // Downsampling by 2.
+  int16_t tmp_out;
+
+  // All-pass filtering upper branch.
+  AllPassFilter(&data_in[0], half_length, kAllPassCoefsQ15[0], upper_state,
+                hp_data_out);
+
+  // All-pass filtering lower branch.
+  AllPassFilter(&data_in[1], half_length, kAllPassCoefsQ15[1], lower_state,
+                lp_data_out);
+
+  // Make LP and HP signals.
+  for (i = 0; i < half_length; i++) {
+    tmp_out = *hp_data_out;
+    *hp_data_out++ -= *lp_data_out;
+    *lp_data_out++ += tmp_out;
+  }
+}
+
+// Calculates the energy of `data_in` in dB, and also updates an overall
+// `total_energy` if necessary.
+//
+// - data_in      [i]   : Input audio data for energy calculation.
+// - data_length  [i]   : Length of input data.
+// - offset       [i]   : Offset value added to `log_energy`.
+// - total_energy [i/o] : An external energy updated with the energy of
+//                        `data_in`.
+//                        NOTE: `total_energy` is only updated if
+//                        `total_energy` <= `kMinEnergy`.
+// - log_energy   [o]   : 10 * log10("energy of `data_in`") given in Q4.
+static void LogOfEnergy(const int16_t* data_in, size_t data_length,
+                        int16_t offset, int16_t* total_energy,
+                        int16_t* log_energy) {
+  // `tot_rshifts` accumulates the number of right shifts performed on `energy`.
+  int tot_rshifts = 0;
+  // The `energy` will be normalized to 15 bits. We use unsigned integer because
+  // we eventually will mask out the fractional part.
+  uint32_t energy = 0;
+
+  RTC_DCHECK(data_in);
+  RTC_DCHECK_GT(data_length, 0);
+
+  energy = (uint32_t) WebRtcSpl_Energy((int16_t*) data_in, data_length,
+                                       &tot_rshifts);
+
+  if (energy != 0) {
+    // By construction, normalizing to 15 bits is equivalent with 17 leading
+    // zeros of an unsigned 32 bit value.
+    int normalizing_rshifts = 17 - WebRtcSpl_NormU32(energy);
+    // In a 15 bit representation the leading bit is 2^14. log2(2^14) in Q10 is
+    // (14 << 10), which is what we initialize `log2_energy` with. For a more
+    // detailed derivations, see below.
+    int16_t log2_energy = kLogEnergyIntPart;
+
+    tot_rshifts += normalizing_rshifts;
+    // Normalize `energy` to 15 bits.
+    // `tot_rshifts` is now the total number of right shifts performed on
+    // `energy` after normalization. This means that `energy` is in
+    // Q(-tot_rshifts).
+    if (normalizing_rshifts < 0) {
+      energy <<= -normalizing_rshifts;
+    } else {
+      energy >>= normalizing_rshifts;
+    }
+
+    // Calculate the energy of `data_in` in dB, in Q4.
+    //
+    // 10 * log10("true energy") in Q4 = 2^4 * 10 * log10("true energy") =
+    // 160 * log10(`energy` * 2^`tot_rshifts`) =
+    // 160 * log10(2) * log2(`energy` * 2^`tot_rshifts`) =
+    // 160 * log10(2) * (log2(`energy`) + log2(2^`tot_rshifts`)) =
+    // (160 * log10(2)) * (log2(`energy`) + `tot_rshifts`) =
+    // `kLogConst` * (`log2_energy` + `tot_rshifts`)
+    //
+    // We know by construction that `energy` is normalized to 15 bits. Hence,
+    // `energy` = 2^14 + frac_Q15, where frac_Q15 is a fractional part in Q15.
+    // Further, we'd like `log2_energy` in Q10
+    // log2(`energy`) in Q10 = 2^10 * log2(2^14 + frac_Q15) =
+    // 2^10 * log2(2^14 * (1 + frac_Q15 * 2^-14)) =
+    // 2^10 * (14 + log2(1 + frac_Q15 * 2^-14)) ~=
+    // (14 << 10) + 2^10 * (frac_Q15 * 2^-14) =
+    // (14 << 10) + (frac_Q15 * 2^-4) = (14 << 10) + (frac_Q15 >> 4)
+    //
+    // Note that frac_Q15 = (`energy` & 0x00003FFF)
+
+    // Calculate and add the fractional part to `log2_energy`.
+    log2_energy += (int16_t) ((energy & 0x00003FFF) >> 4);
+
+    // `kLogConst` is in Q9, `log2_energy` in Q10 and `tot_rshifts` in Q0.
+    // Note that we in our derivation above have accounted for an output in Q4.
+    *log_energy = (int16_t)(((kLogConst * log2_energy) >> 19) +
+        ((tot_rshifts * kLogConst) >> 9));
+
+    if (*log_energy < 0) {
+      *log_energy = 0;
+    }
+  } else {
+    *log_energy = offset;
+    return;
+  }
+
+  *log_energy += offset;
+
+  // Update the approximate `total_energy` with the energy of `data_in`, if
+  // `total_energy` has not exceeded `kMinEnergy`. `total_energy` is used as an
+  // energy indicator in WebRtcVad_GmmProbability() in vad_core.c.
+  if (*total_energy <= kMinEnergy) {
+    if (tot_rshifts >= 0) {
+      // We know by construction that the `energy` > `kMinEnergy` in Q0, so add
+      // an arbitrary value such that `total_energy` exceeds `kMinEnergy`.
+      *total_energy += kMinEnergy + 1;
+    } else {
+      // By construction `energy` is represented by 15 bits, hence any number of
+      // right shifted `energy` will fit in an int16_t. In addition, adding the
+      // value to `total_energy` is wrap around safe as long as
+      // `kMinEnergy` < 8192.
+      *total_energy += (int16_t) (energy >> -tot_rshifts);  // Q0.
+    }
+  }
+}
+
+int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
+                                    size_t data_length, int16_t* features) {
+  int16_t total_energy = 0;
+  // We expect `data_length` to be 80, 160 or 240 samples, which corresponds to
+  // 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
+  // have at most 120 samples after the first split and at most 60 samples after
+  // the second split.
+  int16_t hp_120[120], lp_120[120];
+  int16_t hp_60[60], lp_60[60];
+  const size_t half_data_length = data_length >> 1;
+  size_t length = half_data_length;  // `data_length` / 2, corresponds to
+                                     // bandwidth = 2000 Hz after downsampling.
+
+  // Initialize variables for the first SplitFilter().
+  int frequency_band = 0;
+  const int16_t* in_ptr = data_in;  // [0 - 4000] Hz.
+  int16_t* hp_out_ptr = hp_120;  // [2000 - 4000] Hz.
+  int16_t* lp_out_ptr = lp_120;  // [0 - 2000] Hz.
+
+  RTC_DCHECK_LE(data_length, 240);
+  RTC_DCHECK_LT(4, kNumChannels - 1);  // Checking maximum `frequency_band`.
+
+  // Split at 2000 Hz and downsample.
+  SplitFilter(in_ptr, data_length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // For the upper band (2000 Hz - 4000 Hz) split at 3000 Hz and downsample.
+  frequency_band = 1;
+  in_ptr = hp_120;  // [2000 - 4000] Hz.
+  hp_out_ptr = hp_60;  // [3000 - 4000] Hz.
+  lp_out_ptr = lp_60;  // [2000 - 3000] Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 3000 Hz - 4000 Hz.
+  length >>= 1;  // `data_length` / 4 <=> bandwidth = 1000 Hz.
+
+  LogOfEnergy(hp_60, length, kOffsetVector[5], &total_energy, &features[5]);
+
+  // Energy in 2000 Hz - 3000 Hz.
+  LogOfEnergy(lp_60, length, kOffsetVector[4], &total_energy, &features[4]);
+
+  // For the lower band (0 Hz - 2000 Hz) split at 1000 Hz and downsample.
+  frequency_band = 2;
+  in_ptr = lp_120;  // [0 - 2000] Hz.
+  hp_out_ptr = hp_60;  // [1000 - 2000] Hz.
+  lp_out_ptr = lp_60;  // [0 - 1000] Hz.
+  length = half_data_length;  // `data_length` / 2 <=> bandwidth = 2000 Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 1000 Hz - 2000 Hz.
+  length >>= 1;  // `data_length` / 4 <=> bandwidth = 1000 Hz.
+  LogOfEnergy(hp_60, length, kOffsetVector[3], &total_energy, &features[3]);
+
+  // For the lower band (0 Hz - 1000 Hz) split at 500 Hz and downsample.
+  frequency_band = 3;
+  in_ptr = lp_60;  // [0 - 1000] Hz.
+  hp_out_ptr = hp_120;  // [500 - 1000] Hz.
+  lp_out_ptr = lp_120;  // [0 - 500] Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 500 Hz - 1000 Hz.
+  length >>= 1;  // `data_length` / 8 <=> bandwidth = 500 Hz.
+  LogOfEnergy(hp_120, length, kOffsetVector[2], &total_energy, &features[2]);
+
+  // For the lower band (0 Hz - 500 Hz) split at 250 Hz and downsample.
+  frequency_band = 4;
+  in_ptr = lp_120;  // [0 - 500] Hz.
+  hp_out_ptr = hp_60;  // [250 - 500] Hz.
+  lp_out_ptr = lp_60;  // [0 - 250] Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 250 Hz - 500 Hz.
+  length >>= 1;  // `data_length` / 16 <=> bandwidth = 250 Hz.
+  LogOfEnergy(hp_60, length, kOffsetVector[1], &total_energy, &features[1]);
+
+  // Remove 0 Hz - 80 Hz, by high pass filtering the lower band.
+  HighPassFilter(lp_60, length, self->hp_filter_state, hp_120);
+
+  // Energy in 80 Hz - 250 Hz.
+  LogOfEnergy(hp_120, length, kOffsetVector[0], &total_energy, &features[0]);
+
+  return total_energy;
+}
diff --git a/third_party/libwebrtc/common_audio/vad/vad_filterbank.h b/third_party/libwebrtc/common_audio/vad/vad_filterbank.h
new file mode 100644
index 0000000000..205eac832c
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_filterbank.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+/*
+ * This file includes feature calculating functionality used in vad_core.c.
+ */
+
+#ifndef COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+#define COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+
+#include "common_audio/vad/vad_core.h"
+
+// Takes `data_length` samples of `data_in` and calculates the logarithm of the
+// energy of each of the `kNumChannels` = 6 frequency bands used by the VAD:
+//        80 Hz - 250 Hz
+//        250 Hz - 500 Hz
+//        500 Hz - 1000 Hz
+//        1000 Hz - 2000 Hz
+//        2000 Hz - 3000 Hz
+//        3000 Hz - 4000 Hz
+//
+// The values are given in Q4 and written to `features`. Further, an approximate
+// overall energy is returned. The return value is used in
+// WebRtcVad_GmmProbability() as a signal indicator, hence it is arbitrary above
+// the threshold `kMinEnergy`.
+//
+// - self         [i/o] : State information of the VAD.
+// - data_in      [i]   : Input audio data, for feature extraction.
+// - data_length  [i]   : Audio data size, in number of samples.
+// - features     [o]   : 10 * log10(energy in each frequency band), Q4.
+// - returns            : Total energy of the signal (NOTE! This value is not
+//                        exact. It is only used in a comparison.)
+int16_t WebRtcVad_CalculateFeatures(VadInstT* self,
+                                    const int16_t* data_in,
+                                    size_t data_length,
+                                    int16_t* features);
+
+#endif  // COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
diff --git a/third_party/libwebrtc/common_audio/vad/vad_filterbank_unittest.cc b/third_party/libwebrtc/common_audio/vad/vad_filterbank_unittest.cc
new file mode 100644
index 0000000000..51d8d0fefd
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_filterbank_unittest.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2012 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 <stdlib.h>
+
+#include "common_audio/vad/vad_unittest.h"
+#include "test/gtest.h"
+
+extern "C" {
+#include "common_audio/vad/vad_core.h"
+#include "common_audio/vad/vad_filterbank.h"
+}
+
+namespace webrtc {
+namespace test {
+
+const int kNumValidFrameLengths = 3;
+
+TEST_F(VadTest, vad_filterbank) {
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+  static const int16_t kReference[kNumValidFrameLengths] = {48, 11, 11};
+  static const int16_t kFeatures[kNumValidFrameLengths * kNumChannels] = {
+      1213, 759,  587,  462,  434,  272,  1479, 1385, 1291,
+      1200, 1103, 1099, 1732, 1692, 1681, 1629, 1436, 1436};
+  static const int16_t kOffsetVector[kNumChannels] = {368, 368, 272,
+                                                      176, 176, 176};
+  int16_t features[kNumChannels];
+
+  // Construct a speech signal that will trigger the VAD in all modes. It is
+  // known that (i * i) will wrap around, but that doesn't matter in this case.
+  int16_t speech[kMaxFrameLength];
+  for (size_t i = 0; i < kMaxFrameLength; ++i) {
+    speech[i] = static_cast<int16_t>(i * i);
+  }
+
+  int frame_length_index = 0;
+  ASSERT_EQ(0, WebRtcVad_InitCore(self));
+  for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+    if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+      EXPECT_EQ(kReference[frame_length_index],
+                WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
+                                            features));
+      for (int k = 0; k < kNumChannels; ++k) {
+        EXPECT_EQ(kFeatures[k + frame_length_index * kNumChannels],
+                  features[k]);
+      }
+      frame_length_index++;
+    }
+  }
+  EXPECT_EQ(kNumValidFrameLengths, frame_length_index);
+
+  // Verify that all zeros in gives kOffsetVector out.
+  memset(speech, 0, sizeof(speech));
+  ASSERT_EQ(0, WebRtcVad_InitCore(self));
+  for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+    if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
+                                               features));
+      for (int k = 0; k < kNumChannels; ++k) {
+        EXPECT_EQ(kOffsetVector[k], features[k]);
+      }
+    }
+  }
+
+  // Verify that all ones in gives kOffsetVector out. Any other constant input
+  // will have a small impact in the sub bands.
+  for (size_t i = 0; i < kMaxFrameLength; ++i) {
+    speech[i] = 1;
+  }
+  for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+    if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+      ASSERT_EQ(0, WebRtcVad_InitCore(self));
+      EXPECT_EQ(0, WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
+                                               features));
+      for (int k = 0; k < kNumChannels; ++k) {
+        EXPECT_EQ(kOffsetVector[k], features[k]);
+      }
+    }
+  }
+
+  free(self);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/common_audio/vad/vad_gmm.c b/third_party/libwebrtc/common_audio/vad/vad_gmm.c
new file mode 100644
index 0000000000..4a7fe67d09
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_gmm.c
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2011 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 "common_audio/vad/vad_gmm.h"
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+static const int32_t kCompVar = 22005;
+static const int16_t kLog2Exp = 5909;  // log2(exp(1)) in Q12.
+
+// For a normal distribution, the probability of `input` is calculated and
+// returned (in Q20). The formula for normal distributed probability is
+//
+// 1 / s * exp(-(x - m)^2 / (2 * s^2))
+//
+// where the parameters are given in the following Q domains:
+// m = `mean` (Q7)
+// s = `std` (Q7)
+// x = `input` (Q4)
+// in addition to the probability we output `delta` (in Q11) used when updating
+// the noise/speech model.
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+                                      int16_t mean,
+                                      int16_t std,
+                                      int16_t* delta) {
+  int16_t tmp16, inv_std, inv_std2, exp_value = 0;
+  int32_t tmp32;
+
+  // Calculate `inv_std` = 1 / s, in Q10.
+  // 131072 = 1 in Q17, and (`std` >> 1) is for rounding instead of truncation.
+  // Q-domain: Q17 / Q7 = Q10.
+  tmp32 = (int32_t) 131072 + (int32_t) (std >> 1);
+  inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std);
+
+  // Calculate `inv_std2` = 1 / s^2, in Q14.
+  tmp16 = (inv_std >> 2);  // Q10 -> Q8.
+  // Q-domain: (Q8 * Q8) >> 2 = Q14.
+  inv_std2 = (int16_t)((tmp16 * tmp16) >> 2);
+  // TODO(bjornv): Investigate if changing to
+  // inv_std2 = (int16_t)((inv_std * inv_std) >> 6);
+  // gives better accuracy.
+
+  tmp16 = (input << 3);  // Q4 -> Q7
+  tmp16 = tmp16 - mean;  // Q7 - Q7 = Q7
+
+  // To be used later, when updating noise/speech model.
+  // `delta` = (x - m) / s^2, in Q11.
+  // Q-domain: (Q14 * Q7) >> 10 = Q11.
+  *delta = (int16_t)((inv_std2 * tmp16) >> 10);
+
+  // Calculate the exponent `tmp32` = (x - m)^2 / (2 * s^2), in Q10. Replacing
+  // division by two with one shift.
+  // Q-domain: (Q11 * Q7) >> 8 = Q10.
+  tmp32 = (*delta * tmp16) >> 9;
+
+  // If the exponent is small enough to give a non-zero probability we calculate
+  // `exp_value` ~= exp(-(x - m)^2 / (2 * s^2))
+  //             ~= exp2(-log2(exp(1)) * `tmp32`).
+  if (tmp32 < kCompVar) {
+    // Calculate `tmp16` = log2(exp(1)) * `tmp32`, in Q10.
+    // Q-domain: (Q12 * Q10) >> 12 = Q10.
+    tmp16 = (int16_t)((kLog2Exp * tmp32) >> 12);
+    tmp16 = -tmp16;
+    exp_value = (0x0400 | (tmp16 & 0x03FF));
+    tmp16 ^= 0xFFFF;
+    tmp16 >>= 10;
+    tmp16 += 1;
+    // Get `exp_value` = exp(-`tmp32`) in Q10.
+    exp_value >>= tmp16;
+  }
+
+  // Calculate and return (1 / s) * exp(-(x - m)^2 / (2 * s^2)), in Q20.
+  // Q-domain: Q10 * Q10 = Q20.
+  return inv_std * exp_value;
+}
diff --git a/third_party/libwebrtc/common_audio/vad/vad_gmm.h b/third_party/libwebrtc/common_audio/vad/vad_gmm.h
new file mode 100644
index 0000000000..ada5189756
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_gmm.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+// Gaussian probability calculations internally used in vad_core.c.
+
+#ifndef COMMON_AUDIO_VAD_VAD_GMM_H_
+#define COMMON_AUDIO_VAD_VAD_GMM_H_
+
+#include <stdint.h>
+
+// Calculates the probability for `input`, given that `input` comes from a
+// normal distribution with mean and standard deviation (`mean`, `std`).
+//
+// Inputs:
+//      - input         : input sample in Q4.
+//      - mean          : mean input in the statistical model, Q7.
+//      - std           : standard deviation, Q7.
+//
+// Output:
+//
+//      - delta         : input used when updating the model, Q11.
+//                        `delta` = (`input` - `mean`) / `std`^2.
+//
+// Return:
+//   (probability for `input`) =
+//    1 / `std` * exp(-(`input` - `mean`)^2 / (2 * `std`^2));
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+                                      int16_t mean,
+                                      int16_t std,
+                                      int16_t* delta);
+
+#endif  // COMMON_AUDIO_VAD_VAD_GMM_H_
diff --git a/third_party/libwebrtc/common_audio/vad/vad_gmm_unittest.cc b/third_party/libwebrtc/common_audio/vad/vad_gmm_unittest.cc
new file mode 100644
index 0000000000..be61f7f971
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_gmm_unittest.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 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 "common_audio/vad/vad_unittest.h"
+#include "test/gtest.h"
+
+extern "C" {
+#include "common_audio/vad/vad_gmm.h"
+}
+
+namespace webrtc {
+namespace test {
+
+TEST_F(VadTest, vad_gmm) {
+  int16_t delta = 0;
+  // Input value at mean.
+  EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(0, 0, 128, &delta));
+  EXPECT_EQ(0, delta);
+  EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(16, 128, 128, &delta));
+  EXPECT_EQ(0, delta);
+  EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(-16, -128, 128, &delta));
+  EXPECT_EQ(0, delta);
+
+  // Largest possible input to give non-zero probability.
+  EXPECT_EQ(1024, WebRtcVad_GaussianProbability(59, 0, 128, &delta));
+  EXPECT_EQ(7552, delta);
+  EXPECT_EQ(1024, WebRtcVad_GaussianProbability(75, 128, 128, &delta));
+  EXPECT_EQ(7552, delta);
+  EXPECT_EQ(1024, WebRtcVad_GaussianProbability(-75, -128, 128, &delta));
+  EXPECT_EQ(-7552, delta);
+
+  // Too large input, should give zero probability.
+  EXPECT_EQ(0, WebRtcVad_GaussianProbability(105, 0, 128, &delta));
+  EXPECT_EQ(13440, delta);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/common_audio/vad/vad_sp.c b/third_party/libwebrtc/common_audio/vad/vad_sp.c
new file mode 100644
index 0000000000..3d24cf64b3
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_sp.c
@@ -0,0 +1,176 @@
+/*
+ *  Copyright (c) 2012 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 "common_audio/vad/vad_sp.h"
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/vad/vad_core.h"
+
+// Allpass filter coefficients, upper and lower, in Q13.
+// Upper: 0.64, Lower: 0.17.
+static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 };  // Q13.
+static const int16_t kSmoothingDown = 6553;  // 0.2 in Q15.
+static const int16_t kSmoothingUp = 32439;  // 0.99 in Q15.
+
+// TODO(bjornv): Move this function to vad_filterbank.c.
+// Downsampling filter based on splitting filter and allpass functions.
+void WebRtcVad_Downsampling(const int16_t* signal_in,
+                            int16_t* signal_out,
+                            int32_t* filter_state,
+                            size_t in_length) {
+  int16_t tmp16_1 = 0, tmp16_2 = 0;
+  int32_t tmp32_1 = filter_state[0];
+  int32_t tmp32_2 = filter_state[1];
+  size_t n = 0;
+  // Downsampling by 2 gives half length.
+  size_t half_length = (in_length >> 1);
+
+  // Filter coefficients in Q13, filter state in Q0.
+  for (n = 0; n < half_length; n++) {
+    // All-pass filtering upper branch.
+    tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
+        ((kAllPassCoefsQ13[0] * *signal_in) >> 14));
+    *signal_out = tmp16_1;
+    tmp32_1 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[0] * tmp16_1) >> 12);
+
+    // All-pass filtering lower branch.
+    tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
+        ((kAllPassCoefsQ13[1] * *signal_in) >> 14));
+    *signal_out++ += tmp16_2;
+    tmp32_2 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[1] * tmp16_2) >> 12);
+  }
+  // Store the filter states.
+  filter_state[0] = tmp32_1;
+  filter_state[1] = tmp32_2;
+}
+
+// Inserts `feature_value` into `low_value_vector`, if it is one of the 16
+// smallest values the last 100 frames. Then calculates and returns the median
+// of the five smallest values.
+int16_t WebRtcVad_FindMinimum(VadInstT* self,
+                              int16_t feature_value,
+                              int channel) {
+  int i = 0, j = 0;
+  int position = -1;
+  // Offset to beginning of the 16 minimum values in memory.
+  const int offset = (channel << 4);
+  int16_t current_median = 1600;
+  int16_t alpha = 0;
+  int32_t tmp32 = 0;
+  // Pointer to memory for the 16 minimum values and the age of each value of
+  // the `channel`.
+  int16_t* age = &self->index_vector[offset];
+  int16_t* smallest_values = &self->low_value_vector[offset];
+
+  RTC_DCHECK_LT(channel, kNumChannels);
+
+  // Each value in `smallest_values` is getting 1 loop older. Update `age`, and
+  // remove old values.
+  for (i = 0; i < 16; i++) {
+    if (age[i] != 100) {
+      age[i]++;
+    } else {
+      // Too old value. Remove from memory and shift larger values downwards.
+      for (j = i; j < 15; j++) {
+        smallest_values[j] = smallest_values[j + 1];
+        age[j] = age[j + 1];
+      }
+      age[15] = 101;
+      smallest_values[15] = 10000;
+    }
+  }
+
+  // Check if `feature_value` is smaller than any of the values in
+  // `smallest_values`. If so, find the `position` where to insert the new value
+  // (`feature_value`).
+  if (feature_value < smallest_values[7]) {
+    if (feature_value < smallest_values[3]) {
+      if (feature_value < smallest_values[1]) {
+        if (feature_value < smallest_values[0]) {
+          position = 0;
+        } else {
+          position = 1;
+        }
+      } else if (feature_value < smallest_values[2]) {
+        position = 2;
+      } else {
+        position = 3;
+      }
+    } else if (feature_value < smallest_values[5]) {
+      if (feature_value < smallest_values[4]) {
+        position = 4;
+      } else {
+        position = 5;
+      }
+    } else if (feature_value < smallest_values[6]) {
+      position = 6;
+    } else {
+      position = 7;
+    }
+  } else if (feature_value < smallest_values[15]) {
+    if (feature_value < smallest_values[11]) {
+      if (feature_value < smallest_values[9]) {
+        if (feature_value < smallest_values[8]) {
+          position = 8;
+        } else {
+          position = 9;
+        }
+      } else if (feature_value < smallest_values[10]) {
+        position = 10;
+      } else {
+        position = 11;
+      }
+    } else if (feature_value < smallest_values[13]) {
+      if (feature_value < smallest_values[12]) {
+        position = 12;
+      } else {
+        position = 13;
+      }
+    } else if (feature_value < smallest_values[14]) {
+      position = 14;
+    } else {
+      position = 15;
+    }
+  }
+
+  // If we have detected a new small value, insert it at the correct position
+  // and shift larger values up.
+  if (position > -1) {
+    for (i = 15; i > position; i--) {
+      smallest_values[i] = smallest_values[i - 1];
+      age[i] = age[i - 1];
+    }
+    smallest_values[position] = feature_value;
+    age[position] = 1;
+  }
+
+  // Get `current_median`.
+  if (self->frame_counter > 2) {
+    current_median = smallest_values[2];
+  } else if (self->frame_counter > 0) {
+    current_median = smallest_values[0];
+  }
+
+  // Smooth the median value.
+  if (self->frame_counter > 0) {
+    if (current_median < self->mean_value[channel]) {
+      alpha = kSmoothingDown;  // 0.2 in Q15.
+    } else {
+      alpha = kSmoothingUp;  // 0.99 in Q15.
+    }
+  }
+  tmp32 = (alpha + 1) * self->mean_value[channel];
+  tmp32 += (WEBRTC_SPL_WORD16_MAX - alpha) * current_median;
+  tmp32 += 16384;
+  self->mean_value[channel] = (int16_t) (tmp32 >> 15);
+
+  return self->mean_value[channel];
+}
diff --git a/third_party/libwebrtc/common_audio/vad/vad_sp.h b/third_party/libwebrtc/common_audio/vad/vad_sp.h
new file mode 100644
index 0000000000..89138c57cf
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_sp.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+// This file includes specific signal processing tools used in vad_core.c.
+
+#ifndef COMMON_AUDIO_VAD_VAD_SP_H_
+#define COMMON_AUDIO_VAD_VAD_SP_H_
+
+#include "common_audio/vad/vad_core.h"
+
+// Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
+//
+// Inputs:
+//      - signal_in     : Input signal.
+//      - in_length     : Length of input signal in samples.
+//
+// Input & Output:
+//      - filter_state  : Current filter states of the two all-pass filters. The
+//                        `filter_state` is updated after all samples have been
+//                        processed.
+//
+// Output:
+//      - signal_out    : Downsampled signal (of length `in_length` / 2).
+void WebRtcVad_Downsampling(const int16_t* signal_in,
+                            int16_t* signal_out,
+                            int32_t* filter_state,
+                            size_t in_length);
+
+// Updates and returns the smoothed feature minimum. As minimum we use the
+// median of the five smallest feature values in a 100 frames long window.
+// As long as `handle->frame_counter` is zero, that is, we haven't received any
+// "valid" data, FindMinimum() outputs the default value of 1600.
+//
+// Inputs:
+//      - feature_value : New feature value to update with.
+//      - channel       : Channel number.
+//
+// Input & Output:
+//      - handle        : State information of the VAD.
+//
+// Returns:
+//                      : Smoothed minimum value for a moving window.
+int16_t WebRtcVad_FindMinimum(VadInstT* handle,
+                              int16_t feature_value,
+                              int channel);
+
+#endif  // COMMON_AUDIO_VAD_VAD_SP_H_
diff --git a/third_party/libwebrtc/common_audio/vad/vad_sp_unittest.cc b/third_party/libwebrtc/common_audio/vad/vad_sp_unittest.cc
new file mode 100644
index 0000000000..bf208af3e1
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_sp_unittest.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2012 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 <stdlib.h>
+
+#include "common_audio/vad/vad_unittest.h"
+#include "test/gtest.h"
+
+extern "C" {
+#include "common_audio/vad/vad_core.h"
+#include "common_audio/vad/vad_sp.h"
+}
+
+namespace webrtc {
+namespace test {
+
+TEST_F(VadTest, vad_sp) {
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+  const size_t kMaxFrameLenSp = 960;  // Maximum frame length in this unittest.
+  int16_t zeros[kMaxFrameLenSp] = {0};
+  int32_t state[2] = {0};
+  int16_t data_in[kMaxFrameLenSp];
+  int16_t data_out[kMaxFrameLenSp];
+
+  // We expect the first value to be 1600 as long as `frame_counter` is zero,
+  // which is true for the first iteration.
+  static const int16_t kReferenceMin[32] = {
+      1600, 720, 509, 512, 532, 552,  570, 588, 606, 624, 642,
+      659,  675, 691, 707, 723, 1600, 544, 502, 522, 542, 561,
+      579,  597, 615, 633, 651, 667,  683, 699, 715, 731};
+
+  // Construct a speech signal that will trigger the VAD in all modes. It is
+  // known that (i * i) will wrap around, but that doesn't matter in this case.
+  for (size_t i = 0; i < kMaxFrameLenSp; ++i) {
+    data_in[i] = static_cast<int16_t>(i * i);
+  }
+  // Input values all zeros, expect all zeros out.
+  WebRtcVad_Downsampling(zeros, data_out, state, kMaxFrameLenSp);
+  EXPECT_EQ(0, state[0]);
+  EXPECT_EQ(0, state[1]);
+  for (size_t i = 0; i < kMaxFrameLenSp / 2; ++i) {
+    EXPECT_EQ(0, data_out[i]);
+  }
+  // Make a simple non-zero data test.
+  WebRtcVad_Downsampling(data_in, data_out, state, kMaxFrameLenSp);
+  EXPECT_EQ(207, state[0]);
+  EXPECT_EQ(2270, state[1]);
+
+  ASSERT_EQ(0, WebRtcVad_InitCore(self));
+  // TODO(bjornv): Replace this part of the test with taking values from an
+  // array and calculate the reference value here. Make sure the values are not
+  // ordered.
+  for (int16_t i = 0; i < 16; ++i) {
+    int16_t value = 500 * (i + 1);
+    for (int j = 0; j < kNumChannels; ++j) {
+      // Use values both above and below initialized value.
+      EXPECT_EQ(kReferenceMin[i], WebRtcVad_FindMinimum(self, value, j));
+      EXPECT_EQ(kReferenceMin[i + 16], WebRtcVad_FindMinimum(self, 12000, j));
+    }
+    self->frame_counter++;
+  }
+
+  free(self);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/common_audio/vad/vad_unittest.cc b/third_party/libwebrtc/common_audio/vad/vad_unittest.cc
new file mode 100644
index 0000000000..c54014efce
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_unittest.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2012 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 "common_audio/vad/vad_unittest.h"
+
+#include <stdlib.h>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+VadTest::VadTest() {}
+
+void VadTest::SetUp() {}
+
+void VadTest::TearDown() {}
+
+// Returns true if the rate and frame length combination is valid.
+bool VadTest::ValidRatesAndFrameLengths(int rate, size_t frame_length) {
+  if (rate == 8000) {
+    if (frame_length == 80 || frame_length == 160 || frame_length == 240) {
+      return true;
+    }
+    return false;
+  } else if (rate == 16000) {
+    if (frame_length == 160 || frame_length == 320 || frame_length == 480) {
+      return true;
+    }
+    return false;
+  } else if (rate == 32000) {
+    if (frame_length == 320 || frame_length == 640 || frame_length == 960) {
+      return true;
+    }
+    return false;
+  } else if (rate == 48000) {
+    if (frame_length == 480 || frame_length == 960 || frame_length == 1440) {
+      return true;
+    }
+    return false;
+  }
+
+  return false;
+}
+
+namespace webrtc {
+namespace test {
+
+TEST_F(VadTest, ApiTest) {
+  // This API test runs through the APIs for all possible valid and invalid
+  // combinations.
+
+  VadInst* handle = WebRtcVad_Create();
+  int16_t zeros[kMaxFrameLength] = {0};
+
+  // Construct a speech signal that will trigger the VAD in all modes. It is
+  // known that (i * i) will wrap around, but that doesn't matter in this case.
+  int16_t speech[kMaxFrameLength];
+  for (size_t i = 0; i < kMaxFrameLength; i++) {
+    speech[i] = static_cast<int16_t>(i * i);
+  }
+
+  // nullptr instance tests
+  EXPECT_EQ(-1, WebRtcVad_Init(nullptr));
+  EXPECT_EQ(-1, WebRtcVad_set_mode(nullptr, kModes[0]));
+  EXPECT_EQ(-1,
+            WebRtcVad_Process(nullptr, kRates[0], speech, kFrameLengths[0]));
+
+  // WebRtcVad_Create()
+  RTC_CHECK(handle);
+
+  // Not initialized tests
+  EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], speech, kFrameLengths[0]));
+  EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[0]));
+
+  // WebRtcVad_Init() test
+  ASSERT_EQ(0, WebRtcVad_Init(handle));
+
+  // WebRtcVad_set_mode() invalid modes tests. Tries smallest supported value
+  // minus one and largest supported value plus one.
+  EXPECT_EQ(-1, WebRtcVad_set_mode(
+                    handle, WebRtcSpl_MinValueW32(kModes, kModesSize) - 1));
+  EXPECT_EQ(-1, WebRtcVad_set_mode(
+                    handle, WebRtcSpl_MaxValueW32(kModes, kModesSize) + 1));
+
+  // WebRtcVad_Process() tests
+  // nullptr as speech pointer
+  EXPECT_EQ(-1,
+            WebRtcVad_Process(handle, kRates[0], nullptr, kFrameLengths[0]));
+  // Invalid sampling rate
+  EXPECT_EQ(-1, WebRtcVad_Process(handle, 9999, speech, kFrameLengths[0]));
+  // All zeros as input should work
+  EXPECT_EQ(0, WebRtcVad_Process(handle, kRates[0], zeros, kFrameLengths[0]));
+  for (size_t k = 0; k < kModesSize; k++) {
+    // Test valid modes
+    EXPECT_EQ(0, WebRtcVad_set_mode(handle, kModes[k]));
+    // Loop through sampling rate and frame length combinations
+    for (size_t i = 0; i < kRatesSize; i++) {
+      for (size_t j = 0; j < kFrameLengthsSize; j++) {
+        if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
+          EXPECT_EQ(1, WebRtcVad_Process(handle, kRates[i], speech,
+                                         kFrameLengths[j]));
+        } else {
+          EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[i], speech,
+                                          kFrameLengths[j]));
+        }
+      }
+    }
+  }
+
+  WebRtcVad_Free(handle);
+}
+
+TEST_F(VadTest, ValidRatesFrameLengths) {
+  // This test verifies valid and invalid rate/frame_length combinations. We
+  // loop through some sampling rates and frame lengths from negative values to
+  // values larger than possible.
+  const int kRates[] = {-8000, -4000, 0,     4000,  8000,  8001,
+                        15999, 16000, 32000, 48000, 48001, 96000};
+
+  const size_t kFrameLengths[] = {0,   80,  81,  159, 160,  240,
+                                  320, 480, 640, 960, 1440, 2000};
+
+  for (size_t i = 0; i < arraysize(kRates); i++) {
+    for (size_t j = 0; j < arraysize(kFrameLengths); j++) {
+      if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
+        EXPECT_EQ(
+            0, WebRtcVad_ValidRateAndFrameLength(kRates[i], kFrameLengths[j]));
+      } else {
+        EXPECT_EQ(
+            -1, WebRtcVad_ValidRateAndFrameLength(kRates[i], kFrameLengths[j]));
+      }
+    }
+  }
+}
+
+// TODO(bjornv): Add a process test, run on file.
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/common_audio/vad/vad_unittest.h b/third_party/libwebrtc/common_audio/vad/vad_unittest.h
new file mode 100644
index 0000000000..ee642063af
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/vad_unittest.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2012 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 COMMON_AUDIO_VAD_VAD_UNITTEST_H_
+#define COMMON_AUDIO_VAD_VAD_UNITTEST_H_
+
+#include <stddef.h>  // size_t
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+// Modes we support
+const int kModes[] = {0, 1, 2, 3};
+const size_t kModesSize = sizeof(kModes) / sizeof(*kModes);
+
+// Rates we support.
+const int kRates[] = {8000, 12000, 16000, 24000, 32000, 48000};
+const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
+
+// Frame lengths we support.
+const size_t kMaxFrameLength = 1440;
+const size_t kFrameLengths[] = {
+    80, 120, 160, 240, 320, 480, 640, 960, kMaxFrameLength};
+const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
+
+}  // namespace test
+}  // namespace webrtc
+
+class VadTest : public ::testing::Test {
+ protected:
+  VadTest();
+  void SetUp() override;
+  void TearDown() override;
+
+  // Returns true if the rate and frame length combination is valid.
+  bool ValidRatesAndFrameLengths(int rate, size_t frame_length);
+};
+
+#endif  // COMMON_AUDIO_VAD_VAD_UNITTEST_H_
diff --git a/third_party/libwebrtc/common_audio/vad/webrtc_vad.c b/third_party/libwebrtc/common_audio/vad/webrtc_vad.c
new file mode 100644
index 0000000000..6dd14d8b55
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/vad/webrtc_vad.c
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2012 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 "common_audio/vad/include/webrtc_vad.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/vad/vad_core.h"
+
+static const int kInitCheck = 42;
+static const int kValidRates[] = { 8000, 16000, 32000, 48000 };
+static const size_t kRatesSize = sizeof(kValidRates) / sizeof(*kValidRates);
+static const int kMaxFrameLengthMs = 30;
+
+VadInst* WebRtcVad_Create(void) {
+  VadInstT* self = (VadInstT*)malloc(sizeof(VadInstT));
+
+  self->init_flag = 0;
+
+  return (VadInst*)self;
+}
+
+void WebRtcVad_Free(VadInst* handle) {
+  free(handle);
+}
+
+// TODO(bjornv): Move WebRtcVad_InitCore() code here.
+int WebRtcVad_Init(VadInst* handle) {
+  // Initialize the core VAD component.
+  return WebRtcVad_InitCore((VadInstT*) handle);
+}
+
+// TODO(bjornv): Move WebRtcVad_set_mode_core() code here.
+int WebRtcVad_set_mode(VadInst* handle, int mode) {
+  VadInstT* self = (VadInstT*) handle;
+
+  if (handle == NULL) {
+    return -1;
+  }
+  if (self->init_flag != kInitCheck) {
+    return -1;
+  }
+
+  return WebRtcVad_set_mode_core(self, mode);
+}
+
+int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
+                      size_t frame_length) {
+  int vad = -1;
+  VadInstT* self = (VadInstT*) handle;
+
+  if (handle == NULL) {
+    return -1;
+  }
+
+  if (self->init_flag != kInitCheck) {
+    return -1;
+  }
+  if (audio_frame == NULL) {
+    return -1;
+  }
+  if (WebRtcVad_ValidRateAndFrameLength(fs, frame_length) != 0) {
+    return -1;
+  }
+
+  if (fs == 48000) {
+      vad = WebRtcVad_CalcVad48khz(self, audio_frame, frame_length);
+  } else if (fs == 32000) {
+    vad = WebRtcVad_CalcVad32khz(self, audio_frame, frame_length);
+  } else if (fs == 16000) {
+    vad = WebRtcVad_CalcVad16khz(self, audio_frame, frame_length);
+  } else if (fs == 8000) {
+    vad = WebRtcVad_CalcVad8khz(self, audio_frame, frame_length);
+  }
+
+  if (vad > 0) {
+    vad = 1;
+  }
+  return vad;
+}
+
+int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length) {
+  int return_value = -1;
+  size_t i;
+  int valid_length_ms;
+  size_t valid_length;
+
+  // We only allow 10, 20 or 30 ms frames. Loop through valid frame rates and
+  // see if we have a matching pair.
+  for (i = 0; i < kRatesSize; i++) {
+    if (kValidRates[i] == rate) {
+      for (valid_length_ms = 10; valid_length_ms <= kMaxFrameLengthMs;
+          valid_length_ms += 10) {
+        valid_length = (size_t)(kValidRates[i] / 1000 * valid_length_ms);
+        if (frame_length == valid_length) {
+          return_value = 0;
+          break;
+        }
+      }
+      break;
+    }
+  }
+
+  return return_value;
+}