diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index e0ed2bb..8501dd9 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -33,6 +33,7 @@
     ":common",
     ":gain_applier",
     ":noise_level_estimator",
+    ":rnn_vad_with_level",
     "..:aec_core",
     "..:apm_logging",
     "..:audio_frame_view",
@@ -41,9 +42,6 @@
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
-    "../vad",
-    "../vad:vad_with_level",
-    "rnn_vad",
   ]
 }
 
@@ -133,6 +131,20 @@
   configs += [ "..:apm_debug_dump" ]
 }
 
+rtc_source_set("rnn_vad_with_level") {
+  sources = [
+    "vad_with_level.cc",
+    "vad_with_level.h",
+  ]
+  deps = [
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../common_audio",
+    "../../../rtc_base:checks",
+    "rnn_vad:lib",
+  ]
+}
+
 rtc_source_set("adaptive_digital_unittests") {
   testonly = true
   configs += [ "..:apm_debug_dump" ]
@@ -155,7 +167,6 @@
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:rtc_base_tests_utils",
-    "../vad:vad_with_level",
   ]
 }
 
diff --git a/modules/audio_processing/agc2/adaptive_agc.cc b/modules/audio_processing/agc2/adaptive_agc.cc
index 45e8853..7b24244 100644
--- a/modules/audio_processing/agc2/adaptive_agc.cc
+++ b/modules/audio_processing/agc2/adaptive_agc.cc
@@ -14,8 +14,8 @@
 #include <numeric>
 
 #include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
-#include "modules/audio_processing/vad/voice_activity_detector.h"
 
 namespace webrtc {
 
@@ -38,17 +38,14 @@
   // frames, and no estimates for other frames. We want to feed all to
   // the level estimator, but only care about the last level it
   // produces.
-  rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results =
+  const VadWithLevel::LevelAndProbability vad_result =
       vad_.AnalyzeFrame(float_frame);
-  for (const auto& vad_result : vad_results) {
-    apm_data_dumper_->DumpRaw("agc2_vad_probability",
-                              vad_result.speech_probability);
-    apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs", vad_result.speech_rms_dbfs);
+  apm_data_dumper_->DumpRaw("agc2_vad_probability",
+                            vad_result.speech_probability);
+  apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs", vad_result.speech_rms_dbfs);
 
-    apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs",
-                              vad_result.speech_peak_dbfs);
-    speech_level_estimator_.UpdateEstimation(vad_result);
-  }
+  apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs", vad_result.speech_peak_dbfs);
+  speech_level_estimator_.UpdateEstimation(vad_result);
 
   const float speech_level_dbfs = speech_level_estimator_.LatestLevelEstimate();
 
@@ -57,7 +54,7 @@
   apm_data_dumper_->DumpRaw("agc2_noise_estimate_dbfs", noise_level_dbfs);
 
   // The gain applier applies the gain.
-  gain_applier_.Process(speech_level_dbfs, noise_level_dbfs, vad_results,
+  gain_applier_.Process(speech_level_dbfs, noise_level_dbfs, vad_result,
                         float_frame);
 }
 
diff --git a/modules/audio_processing/agc2/adaptive_agc.h b/modules/audio_processing/agc2/adaptive_agc.h
index a91aa2a..dabe783 100644
--- a/modules/audio_processing/agc2/adaptive_agc.h
+++ b/modules/audio_processing/agc2/adaptive_agc.h
@@ -16,8 +16,8 @@
 #include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
 #include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
 #include "modules/audio_processing/agc2/noise_level_estimator.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
-#include "modules/audio_processing/vad/vad_with_level.h"
 
 namespace webrtc {
 class ApmDataDumper;
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc b/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
index 20b5a27..f5b6b91 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
@@ -74,7 +74,7 @@
 void AdaptiveDigitalGainApplier::Process(
     float input_level_dbfs,
     float input_noise_level_dbfs,
-    rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results,
+    const VadWithLevel::LevelAndProbability vad_result,
     AudioFrameView<float> float_frame) {
   RTC_DCHECK_GE(input_level_dbfs, -150.f);
   RTC_DCHECK_LE(input_level_dbfs, 0.f);
@@ -85,21 +85,9 @@
       LimitGainByNoise(ComputeGainDb(input_level_dbfs), input_noise_level_dbfs,
                        apm_data_dumper_);
 
-  // TODO(webrtc:7494): Remove this construct. Remove the vectors from
-  // VadWithData after we move to a VAD that outputs an estimate every
-  // kFrameDurationMs ms.
-  //
-  // Forbid increasing the gain when there is no speech. For some
-  // VADs, 'vad_results' has either many or 0 results. If there are 0
-  // results, keep the old flag. If there are many results, and at
-  // least one is confident speech, we allow attenuation.
-  if (!vad_results.empty()) {
-    gain_increase_allowed_ = std::all_of(
-        vad_results.begin(), vad_results.end(),
-        [](const VadWithLevel::LevelAndProbability& vad_result) {
-          return vad_result.speech_probability > kVadConfidenceThreshold;
-        });
-  }
+  // Forbid increasing the gain when there is no speech.
+  gain_increase_allowed_ =
+      vad_result.speech_probability > kVadConfidenceThreshold;
 
   const float gain_change_this_frame_db = ComputeGainChangeThisFrameDb(
       target_gain_db, last_gain_db_, gain_increase_allowed_);
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier.h b/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
index b06c65b..31f87f1 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
@@ -13,8 +13,8 @@
 
 #include "modules/audio_processing/agc2/agc2_common.h"
 #include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
-#include "modules/audio_processing/vad/vad_with_level.h"
 
 namespace webrtc {
 
@@ -24,11 +24,10 @@
  public:
   explicit AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper);
   // Decide what gain to apply.
-  void Process(
-      float input_level_dbfs,
-      float input_noise_level_dbfs,
-      rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results,
-      AudioFrameView<float> float_frame);
+  void Process(float input_level_dbfs,
+               float input_noise_level_dbfs,
+               const VadWithLevel::LevelAndProbability vad_result,
+               AudioFrameView<float> float_frame);
 
  private:
   float last_gain_db_ = kInitialAdaptiveDigitalGainDb;
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc b/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc
index ebb040e..860da00 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc
@@ -33,10 +33,8 @@
 
   for (int i = 0; i < num_iterations; ++i) {
     VectorFloatFrame fake_audio(1, 1, 1.f);
-    gain_applier->Process(
-        input_level_dbfs, kNoNoiseDbfs,
-        rtc::ArrayView<const VadWithLevel::LevelAndProbability>(&vad_data, 1),
-        fake_audio.float_frame_view());
+    gain_applier->Process(input_level_dbfs, kNoNoiseDbfs, vad_data,
+                          fake_audio.float_frame_view());
     gain_linear = fake_audio.float_frame_view().channel(0)[0];
   }
   return gain_linear;
@@ -54,10 +52,8 @@
 
   // Make one call with reasonable audio level values and settings.
   VectorFloatFrame fake_audio(2, 480, 10000.f);
-  gain_applier.Process(
-      -5.0, kNoNoiseDbfs,
-      rtc::ArrayView<const VadWithLevel::LevelAndProbability>(&kVadSpeech, 1),
-      fake_audio.float_frame_view());
+  gain_applier.Process(-5.0, kNoNoiseDbfs, kVadSpeech,
+                       fake_audio.float_frame_view());
 }
 
 // Check that the output is -kHeadroom dBFS.
@@ -107,10 +103,8 @@
   for (int i = 0; i < kNumFramesToAdapt; ++i) {
     SCOPED_TRACE(i);
     VectorFloatFrame fake_audio(1, 1, 1.f);
-    gain_applier.Process(
-        initial_level_dbfs, kNoNoiseDbfs,
-        rtc::ArrayView<const VadWithLevel::LevelAndProbability>(&kVadSpeech, 1),
-        fake_audio.float_frame_view());
+    gain_applier.Process(initial_level_dbfs, kNoNoiseDbfs, kVadSpeech,
+                         fake_audio.float_frame_view());
     float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
     EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
               kMaxChangePerFrameLinear);
@@ -121,10 +115,8 @@
   for (int i = 0; i < kNumFramesToAdapt; ++i) {
     SCOPED_TRACE(i);
     VectorFloatFrame fake_audio(1, 1, 1.f);
-    gain_applier.Process(
-        0.f, kNoNoiseDbfs,
-        rtc::ArrayView<const VadWithLevel::LevelAndProbability>(&kVadSpeech, 1),
-        fake_audio.float_frame_view());
+    gain_applier.Process(0.f, kNoNoiseDbfs, kVadSpeech,
+                         fake_audio.float_frame_view());
     float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
     EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
               kMaxChangePerFrameLinear);
@@ -140,10 +132,8 @@
   constexpr int num_samples = 480;
 
   VectorFloatFrame fake_audio(1, num_samples, 1.f);
-  gain_applier.Process(
-      initial_level_dbfs, kNoNoiseDbfs,
-      rtc::ArrayView<const VadWithLevel::LevelAndProbability>(&kVadSpeech, 1),
-      fake_audio.float_frame_view());
+  gain_applier.Process(initial_level_dbfs, kNoNoiseDbfs, kVadSpeech,
+                       fake_audio.float_frame_view());
   float maximal_difference = 0.f;
   float current_value = 1.f * DbToRatio(kInitialAdaptiveDigitalGainDb);
   for (const auto& x : fake_audio.float_frame_view().channel(0)) {
@@ -172,10 +162,8 @@
 
   for (int i = 0; i < num_initial_frames + num_frames; ++i) {
     VectorFloatFrame fake_audio(1, num_samples, 1.f);
-    gain_applier.Process(
-        initial_level_dbfs, kWithNoiseDbfs,
-        rtc::ArrayView<const VadWithLevel::LevelAndProbability>(&kVadSpeech, 1),
-        fake_audio.float_frame_view());
+    gain_applier.Process(initial_level_dbfs, kWithNoiseDbfs, kVadSpeech,
+                         fake_audio.float_frame_view());
 
     // Wait so that the adaptive gain applier has time to lower the gain.
     if (i > num_initial_frames) {
diff --git a/modules/audio_processing/agc2/adaptive_mode_level_estimator.h b/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
index 9762f1f..186c59b 100644
--- a/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
+++ b/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
@@ -12,7 +12,7 @@
 #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
 
 #include "modules/audio_processing/agc2/saturation_protector.h"
-#include "modules/audio_processing/vad/vad_with_level.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
 
 namespace webrtc {
 class ApmDataDumper;
diff --git a/modules/audio_processing/agc2/agc2_common.h b/modules/audio_processing/agc2/agc2_common.h
index 3ed88a3..7300653 100644
--- a/modules/audio_processing/agc2/agc2_common.h
+++ b/modules/audio_processing/agc2/agc2_common.h
@@ -41,10 +41,10 @@
 // Used in the Level Estimator for deciding when to update the speech
 // level estimate. Also used in the adaptive digital gain applier to
 // decide when to allow target gain reduction.
-constexpr float kVadConfidenceThreshold = 0.9f;
+constexpr float kVadConfidenceThreshold = 0.4f;
 
 // The amount of 'memory' of the Level Estimator. Decides leak factors.
-constexpr size_t kFullBufferSizeMs = 1000;
+constexpr size_t kFullBufferSizeMs = 1600;
 constexpr float kFullBufferLeakFactor = 1.f - 1.f / kFullBufferSizeMs;
 
 constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
@@ -52,7 +52,10 @@
 // Saturation Protector settings.
 constexpr float kInitialSaturationMarginDb = 17.f;
 
-constexpr size_t kPeakEnveloperSuperFrameLengthMs = 500;
+constexpr size_t kPeakEnveloperSuperFrameLengthMs = 400;
+static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,
+              "Full buffer size should be a multiple of super frame length for "
+              "optimal Saturation Protector performance.");
 
 constexpr size_t kPeakEnveloperBufferSize =
     kFullBufferSizeMs / kPeakEnveloperSuperFrameLengthMs + 1;
diff --git a/modules/audio_processing/agc2/saturation_protector.h b/modules/audio_processing/agc2/saturation_protector.h
index d330c15..3a796fa 100644
--- a/modules/audio_processing/agc2/saturation_protector.h
+++ b/modules/audio_processing/agc2/saturation_protector.h
@@ -14,7 +14,7 @@
 #include <array>
 
 #include "modules/audio_processing/agc2/agc2_common.h"
-#include "modules/audio_processing/vad/vad_with_level.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
 
 namespace webrtc {
 
diff --git a/modules/audio_processing/agc2/saturation_protector_unittest.cc b/modules/audio_processing/agc2/saturation_protector_unittest.cc
index 88da2a2..6013e13 100644
--- a/modules/audio_processing/agc2/saturation_protector_unittest.cc
+++ b/modules/audio_processing/agc2/saturation_protector_unittest.cc
@@ -30,6 +30,7 @@
     max_difference =
         std::max(max_difference, std::abs(new_margin - last_margin));
     last_margin = new_margin;
+    saturation_protector->DebugDumpEstimate();
   }
   return max_difference;
 }
@@ -127,6 +128,12 @@
           kLaterSpeechLevelDbfs, &saturation_protector),
       max_difference);
 
+  // The saturation protector expects that the RMS changes roughly
+  // 'kFullBufferSizeMs' after peaks change. This is to account for
+  // delay introduces by the level estimator. Therefore, the input
+  // above is 'normal' and 'expected', and shouldn't influence the
+  // margin by much.
+
   const float total_difference =
       std::abs(saturation_protector.LastMargin() - kInitialSaturationMarginDb);
 
diff --git a/modules/audio_processing/agc2/vad_with_level.cc b/modules/audio_processing/agc2/vad_with_level.cc
new file mode 100644
index 0000000..decfacd
--- /dev/null
+++ b/modules/audio_processing/agc2/vad_with_level.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/vad_with_level.h"
+
+#include <algorithm>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+float ProcessForPeak(AudioFrameView<const float> frame) {
+  float current_max = 0;
+  for (const auto& x : frame.channel(0)) {
+    current_max = std::max(std::fabs(x), current_max);
+  }
+  return current_max;
+}
+
+float ProcessForRms(AudioFrameView<const float> frame) {
+  float rms = 0;
+  for (const auto& x : frame.channel(0)) {
+    rms += x * x;
+  }
+  return sqrt(rms / frame.samples_per_channel());
+}
+}  // namespace
+
+VadWithLevel::VadWithLevel() = default;
+VadWithLevel::~VadWithLevel() = default;
+
+VadWithLevel::LevelAndProbability VadWithLevel::AnalyzeFrame(
+    AudioFrameView<const float> frame) {
+  SetSampleRate(static_cast<int>(frame.samples_per_channel() * 100));
+  std::array<float, rnn_vad::kFrameSize10ms24kHz> work_frame;
+  // Feed the 1st channel to the resampler.
+  resampler_.Resample(frame.channel(0).data(), frame.samples_per_channel(),
+                      work_frame.data(), rnn_vad::kFrameSize10ms24kHz);
+
+  std::array<float, rnn_vad::kFeatureVectorSize> feature_vector;
+
+  const bool is_silence = features_extractor_.CheckSilenceComputeFeatures(
+      work_frame, feature_vector);
+  const float vad_probability =
+      rnn_vad_.ComputeVadProbability(feature_vector, is_silence);
+  return LevelAndProbability(vad_probability,
+                             FloatS16ToDbfs(ProcessForRms(frame)),
+                             FloatS16ToDbfs(ProcessForPeak(frame)));
+}
+
+void VadWithLevel::SetSampleRate(int sample_rate_hz) {
+  // The source number of channels in 1, because we always use the 1st
+  // channel.
+  resampler_.InitializeIfNeeded(sample_rate_hz, rnn_vad::kSampleRate24kHz,
+                                1 /* num_channels */);
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_processing/vad/vad_with_level.h b/modules/audio_processing/agc2/vad_with_level.h
similarity index 60%
rename from modules/audio_processing/vad/vad_with_level.h
rename to modules/audio_processing/agc2/vad_with_level.h
index 9ad4d17..67a00ce 100644
--- a/modules/audio_processing/vad/vad_with_level.h
+++ b/modules/audio_processing/agc2/vad_with_level.h
@@ -8,10 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_
-#define MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
 
 #include "api/array_view.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 
 namespace webrtc {
@@ -28,13 +31,19 @@
     float speech_peak_dbfs = 0;
   };
 
-  // TODO(webrtc:7494): This is a stub. Add implementation.
-  rtc::ArrayView<const LevelAndProbability> AnalyzeFrame(
-      AudioFrameView<const float> frame) {
-    return {nullptr, 0};
-  }
+  VadWithLevel();
+  ~VadWithLevel();
+
+  LevelAndProbability AnalyzeFrame(AudioFrameView<const float> frame);
+
+ private:
+  void SetSampleRate(int sample_rate_hz);
+
+  rnn_vad::RnnBasedVad rnn_vad_;
+  rnn_vad::FeaturesExtractor features_extractor_;
+  PushResampler<float> resampler_;
 };
 
 }  // namespace webrtc
 
-#endif  // MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
diff --git a/modules/audio_processing/vad/BUILD.gn b/modules/audio_processing/vad/BUILD.gn
index 9a57789..ae2a84d 100644
--- a/modules/audio_processing/vad/BUILD.gn
+++ b/modules/audio_processing/vad/BUILD.gn
@@ -44,16 +44,6 @@
   ]
 }
 
-rtc_source_set("vad_with_level") {
-  sources = [
-    "vad_with_level.h",
-  ]
-  deps = [
-    "..:audio_frame_view",
-    "../../../api:array_view",
-  ]
-}
-
 if (rtc_include_tests) {
   rtc_static_library("vad_unittests") {
     testonly = true