diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn
index 0103655ba9..b01b31816b 100644
--- a/webrtc/common_audio/BUILD.gn
+++ b/webrtc/common_audio/BUILD.gn
@@ -21,7 +21,11 @@ source_set("common_audio") {
   sources = [
     "audio_converter.cc",
     "audio_converter.h",
+    "audio_ring_buffer.cc",
+    "audio_ring_buffer.h",
     "audio_util.cc",
+    "blocker.cc",
+    "blocker.h",
     "channel_buffer.cc",
     "channel_buffer.h",
     "fft4g.c",
@@ -31,6 +35,8 @@ source_set("common_audio") {
     "fir_filter_neon.h",
     "fir_filter_sse.h",
     "include/audio_util.h",
+    "lapped_transform.cc",
+    "lapped_transform.h",
     "real_fourier.cc",
     "real_fourier.h",
     "real_fourier_ooura.cc",
@@ -43,6 +49,8 @@ source_set("common_audio") {
     "resampler/resampler.cc",
     "resampler/sinc_resampler.cc",
     "resampler/sinc_resampler.h",
+    "ring_buffer.c",
+    "ring_buffer.h",
     "signal_processing/auto_corr_to_refl_coef.c",
     "signal_processing/auto_correlation.c",
     "signal_processing/complex_fft_tables.h",
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer.cc b/webrtc/common_audio/audio_ring_buffer.cc
similarity index 94%
rename from webrtc/modules/audio_processing/utility/audio_ring_buffer.cc
rename to webrtc/common_audio/audio_ring_buffer.cc
index 73f578f58b..a29e53a61c 100644
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer.cc
+++ b/webrtc/common_audio/audio_ring_buffer.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
+#include "webrtc/common_audio/audio_ring_buffer.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
 
 // This is a simple multi-channel wrapper over the ring_buffer.h C interface.
 
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer.h b/webrtc/common_audio/audio_ring_buffer.h
similarity index 89%
rename from webrtc/modules/audio_processing/utility/audio_ring_buffer.h
rename to webrtc/common_audio/audio_ring_buffer.h
index 8f9758770e..ae825a3cd0 100644
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer.h
+++ b/webrtc/common_audio/audio_ring_buffer.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
+#ifndef WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
+#define WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
 
 #include <stddef.h>
 #include <vector>
@@ -52,4 +52,4 @@ class AudioRingBuffer final {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
+#endif  // WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc b/webrtc/common_audio/audio_ring_buffer_unittest.cc
similarity index 97%
rename from webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc
rename to webrtc/common_audio/audio_ring_buffer_unittest.cc
index c2d5e7a64c..c5c38de56d 100644
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc
+++ b/webrtc/common_audio/audio_ring_buffer_unittest.cc
@@ -9,9 +9,8 @@
  */
 
 #include <memory>
-#include <tuple>
 
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
+#include "webrtc/common_audio/audio_ring_buffer.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_audio/channel_buffer.h"
diff --git a/webrtc/modules/audio_processing/utility/blocker.cc b/webrtc/common_audio/blocker.cc
similarity index 99%
rename from webrtc/modules/audio_processing/utility/blocker.cc
rename to webrtc/common_audio/blocker.cc
index a3661cc156..13432f2e7a 100644
--- a/webrtc/modules/audio_processing/utility/blocker.cc
+++ b/webrtc/common_audio/blocker.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/utility/blocker.h"
+#include "webrtc/common_audio/blocker.h"
 
 #include <string.h>
 
diff --git a/webrtc/modules/audio_processing/utility/blocker.h b/webrtc/common_audio/blocker.h
similarity index 94%
rename from webrtc/modules/audio_processing/utility/blocker.h
rename to webrtc/common_audio/blocker.h
index 7d9bf66e48..edf81d337a 100644
--- a/webrtc/modules/audio_processing/utility/blocker.h
+++ b/webrtc/common_audio/blocker.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
+#ifndef WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
+#define WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
 
 #include <memory>
 
+#include "webrtc/common_audio/audio_ring_buffer.h"
 #include "webrtc/common_audio/channel_buffer.h"
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
 
 namespace webrtc {
 
@@ -121,4 +121,4 @@ class Blocker {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
+#endif  // WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
diff --git a/webrtc/modules/audio_processing/utility/blocker_unittest.cc b/webrtc/common_audio/blocker_unittest.cc
similarity index 99%
rename from webrtc/modules/audio_processing/utility/blocker_unittest.cc
rename to webrtc/common_audio/blocker_unittest.cc
index 3f53f68660..eea3e2516a 100644
--- a/webrtc/modules/audio_processing/utility/blocker_unittest.cc
+++ b/webrtc/common_audio/blocker_unittest.cc
@@ -10,7 +10,7 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_processing/utility/blocker.h"
+#include "webrtc/common_audio/blocker.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/arraysize.h"
diff --git a/webrtc/common_audio/common_audio.gyp b/webrtc/common_audio/common_audio.gyp
index e5a93798b5..57d9f1ca64 100644
--- a/webrtc/common_audio/common_audio.gyp
+++ b/webrtc/common_audio/common_audio.gyp
@@ -31,7 +31,11 @@
       'sources': [
         'audio_converter.cc',
         'audio_converter.h',
+        'audio_ring_buffer.cc',
+        'audio_ring_buffer.h',
         'audio_util.cc',
+        'blocker.cc',
+        'blocker.h',
         'channel_buffer.cc',
         'channel_buffer.h',
         'fft4g.c',
@@ -41,6 +45,8 @@
         'fir_filter_neon.h',
         'fir_filter_sse.h',
         'include/audio_util.h',
+        'lapped_transform.cc',
+        'lapped_transform.h',
         'real_fourier.cc',
         'real_fourier.h',
         'real_fourier_ooura.cc',
@@ -53,6 +59,8 @@
         'resampler/resampler.cc',
         'resampler/sinc_resampler.cc',
         'resampler/sinc_resampler.h',
+        'ring_buffer.c',
+        'ring_buffer.h',
         'signal_processing/include/real_fft.h',
         'signal_processing/include/signal_processing_library.h',
         'signal_processing/include/spl_inl.h',
@@ -232,14 +240,18 @@
           ],
           'sources': [
             'audio_converter_unittest.cc',
+            'audio_ring_buffer_unittest.cc',
             'audio_util_unittest.cc',
+            'blocker_unittest.cc',
             'fir_filter_unittest.cc',
+            'lapped_transform_unittest.cc',
             'real_fourier_unittest.cc',
             'resampler/resampler_unittest.cc',
             'resampler/push_resampler_unittest.cc',
             'resampler/push_sinc_resampler_unittest.cc',
             'resampler/sinusoidal_linear_chirp_source.cc',
             'resampler/sinusoidal_linear_chirp_source.h',
+            'ring_buffer_unittest.cc',
             'signal_processing/real_fft_unittest.cc',
             'signal_processing/signal_processing_unittest.cc',
             'sparse_fir_filter_unittest.cc',
diff --git a/webrtc/modules/audio_processing/utility/lapped_transform.cc b/webrtc/common_audio/lapped_transform.cc
similarity index 98%
rename from webrtc/modules/audio_processing/utility/lapped_transform.cc
rename to webrtc/common_audio/lapped_transform.cc
index cb5496dfc4..0edf586d78 100644
--- a/webrtc/modules/audio_processing/utility/lapped_transform.cc
+++ b/webrtc/common_audio/lapped_transform.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
+#include "webrtc/common_audio/lapped_transform.h"
 
 #include <algorithm>
 #include <cstdlib>
diff --git a/webrtc/modules/audio_processing/utility/lapped_transform.h b/webrtc/common_audio/lapped_transform.h
similarity index 94%
rename from webrtc/modules/audio_processing/utility/lapped_transform.h
rename to webrtc/common_audio/lapped_transform.h
index 1286ecf2d8..832735991b 100644
--- a/webrtc/modules/audio_processing/utility/lapped_transform.h
+++ b/webrtc/common_audio/lapped_transform.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
+#ifndef WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
+#define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
 
 #include <complex>
 #include <memory>
 
+#include "webrtc/common_audio/blocker.h"
 #include "webrtc/common_audio/real_fourier.h"
-#include "webrtc/modules/audio_processing/utility/blocker.h"
 #include "webrtc/system_wrappers/include/aligned_array.h"
 
 namespace webrtc {
@@ -121,4 +121,5 @@ class LappedTransform {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
+#endif  // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
+
diff --git a/webrtc/modules/audio_processing/utility/lapped_transform_unittest.cc b/webrtc/common_audio/lapped_transform_unittest.cc
similarity index 98%
rename from webrtc/modules/audio_processing/utility/lapped_transform_unittest.cc
rename to webrtc/common_audio/lapped_transform_unittest.cc
index f6c8ebd6d3..a78488e326 100644
--- a/webrtc/modules/audio_processing/utility/lapped_transform_unittest.cc
+++ b/webrtc/common_audio/lapped_transform_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
+#include "webrtc/common_audio/lapped_transform.h"
 
 #include <algorithm>
 #include <cmath>
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer.c b/webrtc/common_audio/ring_buffer.c
similarity index 99%
rename from webrtc/modules/audio_processing/utility/ring_buffer.c
rename to webrtc/common_audio/ring_buffer.c
index c71bac1862..60fb5dff20 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer.c
+++ b/webrtc/common_audio/ring_buffer.c
@@ -11,7 +11,7 @@
 // A ring buffer to hold arbitrary data. Provides no thread safety. Unless
 // otherwise specified, functions return 0 on success and -1 on error.
 
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
 
 #include <stddef.h>  // size_t
 #include <stdlib.h>
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer.h b/webrtc/common_audio/ring_buffer.h
similarity index 92%
rename from webrtc/modules/audio_processing/utility/ring_buffer.h
rename to webrtc/common_audio/ring_buffer.h
index a46c262229..4125c48d01 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer.h
+++ b/webrtc/common_audio/ring_buffer.h
@@ -11,8 +11,8 @@
 // A ring buffer to hold arbitrary data. Provides no thread safety. Unless
 // otherwise specified, functions return 0 on success and -1 on error.
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+#ifndef WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
+#define WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
 
 #ifdef __cplusplus
 extern "C" {
@@ -63,4 +63,4 @@ size_t WebRtc_available_write(const RingBuffer* handle);
 }
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+#endif  // WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc b/webrtc/common_audio/ring_buffer_unittest.cc
similarity index 98%
rename from webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc
rename to webrtc/common_audio/ring_buffer_unittest.cc
index 7972657727..92c470a02d 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc
+++ b/webrtc/common_audio/ring_buffer_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
 
 #include <stdlib.h>
 #include <time.h>
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index bf7bdc5d1c..22c904d52f 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -108,21 +108,13 @@ source_set("audio_processing") {
     "transient/wpd_tree.h",
     "typing_detection.cc",
     "typing_detection.h",
-    "utility/audio_ring_buffer.cc",
-    "utility/audio_ring_buffer.h",
     "utility/block_mean_calculator.cc",
     "utility/block_mean_calculator.h",
-    "utility/blocker.cc",
-    "utility/blocker.h",
     "utility/delay_estimator.c",
     "utility/delay_estimator.h",
     "utility/delay_estimator_internal.h",
     "utility/delay_estimator_wrapper.c",
     "utility/delay_estimator_wrapper.h",
-    "utility/lapped_transform.cc",
-    "utility/lapped_transform.h",
-    "utility/ring_buffer.c",
-    "utility/ring_buffer.h",
     "vad/common.h",
     "vad/gmm.cc",
     "vad/gmm.h",
diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc
index 4fe635d2db..e23a79312b 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core.cc
@@ -24,6 +24,9 @@
 #include <stdlib.h>
 #include <string.h>
 
+extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
+}
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
 #include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
@@ -33,7 +36,6 @@ extern "C" {
 #include "webrtc/modules/audio_processing/logging/aec_logging.h"
 extern "C" {
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
 }
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_processing/aec/aec_core_internal.h b/webrtc/modules/audio_processing/aec/aec_core_internal.h
index 1f4f99be63..ea5889f503 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_internal.h
+++ b/webrtc/modules/audio_processing/aec/aec_core_internal.h
@@ -11,14 +11,13 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
 
+extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
+}
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/modules/audio_processing/utility/block_mean_calculator.h"
-extern "C" {
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
-}
-
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.cc b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
index 493068d9bf..32496ca33c 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.cc
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
@@ -21,14 +21,12 @@
 #include <string.h>
 
 extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 }
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/modules/audio_processing/aec/aec_resampler.h"
 #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
-extern "C" {
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
-}
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
index 537ab5d904..b4a6fd8390 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
 
-#include "webrtc/modules/audio_processing/aec/aec_core.h"
 extern "C" {
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
 }
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index 084feb858b..6bf1cf7f3e 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -14,10 +14,10 @@
 #include <stddef.h>
 #include <stdlib.h>
 
+#include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
 #include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
 #include "webrtc/system_wrappers/include/compile_assert_c.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.h b/webrtc/modules/audio_processing/aecm/aecm_core.h
index 9b70f47503..b52bb62d2d 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.h
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.h
@@ -13,9 +13,9 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
 
+#include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/aecm/aecm_defines.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
 #include "webrtc/typedefs.h"
 
 #ifdef _MSC_VER  // visual c++
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
index ea5f2bc8cf..3a8fafa4ec 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_c.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
@@ -14,10 +14,10 @@
 #include <stddef.h>
 #include <stdlib.h>
 
+#include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
 #include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
 #include "webrtc/system_wrappers/include/compile_assert_c.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
@@ -768,3 +768,4 @@ static void ComfortNoise(AecmCore* aecm,
     out[i].imag = WebRtcSpl_AddSatW16(out[i].imag, uImag[i]);
   }
 }
+
diff --git a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
index d58ee3e456..91e6f0e80c 100644
--- a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
+++ b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
@@ -15,9 +15,9 @@
 #endif
 #include <stdlib.h>
 
+#include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/aecm/aecm_core.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
 
 #define BUF_SIZE_FRAMES 50 // buffer size (frames)
 // Maximum length of resampled signal. Must be an integer multiple of frames
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 75da59f3e0..264f3e5bef 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -118,21 +118,13 @@
         'transient/wpd_tree.h',
         'typing_detection.cc',
         'typing_detection.h',
-        'utility/audio_ring_buffer.cc',
-        'utility/audio_ring_buffer.h',
         'utility/block_mean_calculator.cc',
         'utility/block_mean_calculator.h',
-        'utility/blocker.cc',
-        'utility/blocker.h',
         'utility/delay_estimator.c',
         'utility/delay_estimator.h',
         'utility/delay_estimator_internal.h',
         'utility/delay_estimator_wrapper.c',
         'utility/delay_estimator_wrapper.h',
-        'utility/lapped_transform.cc',
-        'utility/lapped_transform.h',
-        'utility/ring_buffer.c',
-        'utility/ring_buffer.h',
         'vad/common.h',
         'vad/gmm.cc',
         'vad/gmm.h',
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
index c1f9188bfc..b8953b0a4f 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
@@ -19,10 +19,11 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/common_audio/lapped_transform.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/beamformer/beamformer.h"
 #include "webrtc/modules/audio_processing/beamformer/complex_matrix.h"
-#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
+
 namespace webrtc {
 
 // Enhances sound sources coming directly in front of a uniform linear array
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
index 728e4cfb70..1413212934 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -16,10 +16,10 @@
 #include <vector>
 
 #include "webrtc/base/swap_queue.h"
+#include "webrtc/common_audio/lapped_transform.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
-#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
 #include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index eb886a9e4a..d5df853905 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -258,12 +258,8 @@
             'audio_processing/transient/transient_suppressor_unittest.cc',
             'audio_processing/transient/wpd_node_unittest.cc',
             'audio_processing/transient/wpd_tree_unittest.cc',
-            'audio_processing/utility/audio_ring_buffer_unittest.cc',
             'audio_processing/utility/block_mean_calculator_unittest.cc',
-            'audio_processing/utility/blocker_unittest.cc',
             'audio_processing/utility/delay_estimator_unittest.cc',
-            'audio_processing/utility/lapped_transform_unittest.cc',
-            'audio_processing/utility/ring_buffer_unittest.cc',
             'audio_processing/vad/gmm_unittest.cc',
             'audio_processing/vad/pitch_based_vad_unittest.cc',
             'audio_processing/vad/pitch_internal_unittest.cc',