diff --git a/modules/audio_processing/aec3/aec3_fft.cc b/modules/audio_processing/aec3/aec3_fft.cc
index 1832101855..d1d4f7da06 100644
--- a/modules/audio_processing/aec3/aec3_fft.cc
+++ b/modules/audio_processing/aec3/aec3_fft.cc
@@ -15,6 +15,7 @@
 #include <iterator>
 
 #include "rtc_base/checks.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
 
 namespace webrtc {
 
@@ -70,8 +71,18 @@ const float kSqrtHanning128[kFftLength] = {
     0.19509032201613f, 0.17096188876030f, 0.14673047445536f, 0.12241067519922f,
     0.09801714032956f, 0.07356456359967f, 0.04906767432742f, 0.02454122852291f};
 
+bool IsSse2Available() {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+  return WebRtc_GetCPUInfo(kSSE2) != 0;
+#else
+  return false;
+#endif
+}
+
 }  // namespace
 
+Aec3Fft::Aec3Fft() : ooura_fft_(IsSse2Available()) {}
+
 // TODO(peah): Change x to be std::array once the rest of the code allows this.
 void Aec3Fft::ZeroPaddedFft(rtc::ArrayView<const float> x,
                             Window window,
diff --git a/modules/audio_processing/aec3/aec3_fft.h b/modules/audio_processing/aec3/aec3_fft.h
index 7a2e024d75..6f7fbe4d0e 100644
--- a/modules/audio_processing/aec3/aec3_fft.h
+++ b/modules/audio_processing/aec3/aec3_fft.h
@@ -28,7 +28,8 @@ class Aec3Fft {
  public:
   enum class Window { kRectangular, kHanning, kSqrtHanning };
 
-  Aec3Fft() = default;
+  Aec3Fft();
+
   // Computes the FFT. Note that both the input and output are modified.
   void Fft(std::array<float, kFftLength>* x, FftData* X) const {
     RTC_DCHECK(x);
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 8d9bb14731..bfef2252c3 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -153,6 +153,7 @@ rtc_library("noise_level_estimator") {
     "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../rtc_base:checks",
     "../../../rtc_base:macromagic",
+    "../../../system_wrappers:cpu_features_api",
   ]
 
   configs += [ "..:apm_debug_dump" ]
diff --git a/modules/audio_processing/agc2/signal_classifier.cc b/modules/audio_processing/agc2/signal_classifier.cc
index 8778c49426..38334f7ec5 100644
--- a/modules/audio_processing/agc2/signal_classifier.cc
+++ b/modules/audio_processing/agc2/signal_classifier.cc
@@ -19,10 +19,19 @@
 #include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/checks.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
 
 namespace webrtc {
 namespace {
 
+bool IsSse2Available() {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+  return WebRtc_GetCPUInfo(kSSE2) != 0;
+#else
+  return false;
+#endif
+}
+
 void RemoveDcLevel(rtc::ArrayView<float> x) {
   RTC_DCHECK_LT(0, x.size());
   float mean = std::accumulate(x.data(), x.data() + x.size(), 0.f);
@@ -109,7 +118,8 @@ void SignalClassifier::FrameExtender::ExtendFrame(
 SignalClassifier::SignalClassifier(ApmDataDumper* data_dumper)
     : data_dumper_(data_dumper),
       down_sampler_(data_dumper_),
-      noise_spectrum_estimator_(data_dumper_) {
+      noise_spectrum_estimator_(data_dumper_),
+      ooura_fft_(IsSse2Available()) {
   Initialize(48000);
 }
 SignalClassifier::~SignalClassifier() {}