diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 200d709e8d..1d14c906f3 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -536,7 +536,6 @@ if (rtc_include_tests) {
     deps = [
       "base:rtc_base",
       "base:rtc_base_tests_utils",
-      "base:rtc_exp_filter",
       "base:rtc_task_queue",
       "p2p:libstunprober",
       "p2p:rtc_p2p",
diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn
index 89b5725138..e0f6b8282f 100644
--- a/webrtc/base/BUILD.gn
+++ b/webrtc/base/BUILD.gn
@@ -22,7 +22,6 @@ group("base") {
   public_deps = [
     ":rtc_base",
     ":rtc_base_approved",
-    ":rtc_exp_filter",
     ":rtc_task_queue",
   ]
   if (is_android) {
@@ -125,6 +124,8 @@ rtc_static_library("rtc_base_approved") {
     "event.h",
     "event_tracer.cc",
     "event_tracer.h",
+    "exp_filter.cc",
+    "exp_filter.h",
     "file.cc",
     "file.h",
     "format_macros.h",
@@ -337,13 +338,6 @@ rtc_static_library("rtc_task_queue") {
   }
 }
 
-rtc_static_library("rtc_exp_filter") {
-  sources = [
-    "exp_filter.cc",
-    "exp_filter.h",
-  ]
-}
-
 config("rtc_base_warnings_config") {
   if (is_win && is_clang) {
     cflags = [
diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn
index bd85e2e5b9..bfed73eeeb 100644
--- a/webrtc/common_audio/BUILD.gn
+++ b/webrtc/common_audio/BUILD.gn
@@ -86,8 +86,6 @@ rtc_static_library("common_audio") {
     "signal_processing/splitting_filter.c",
     "signal_processing/sqrt_of_one_minus_x_squared.c",
     "signal_processing/vector_scaling_operations.c",
-    "smoothing_filter.cc",
-    "smoothing_filter.h",
     "sparse_fir_filter.cc",
     "sparse_fir_filter.h",
     "vad/include/vad.h",
@@ -111,7 +109,6 @@ rtc_static_library("common_audio") {
   ]
 
   deps = [
-    "../base:rtc_exp_filter",
     "../system_wrappers",
   ]
 
@@ -260,7 +257,6 @@ if (rtc_include_tests) {
       "ring_buffer_unittest.cc",
       "signal_processing/real_fft_unittest.cc",
       "signal_processing/signal_processing_unittest.cc",
-      "smoothing_filter_unittest.cc",
       "sparse_fir_filter_unittest.cc",
       "vad/vad_core_unittest.cc",
       "vad/vad_filterbank_unittest.cc",
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 7c8a49644e..9120541041 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -264,6 +264,7 @@ if (rtc_include_tests) {
       "audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc",
       "audio_coding/audio_network_adaptor/mock/mock_controller.h",
       "audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
+      "audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc",
     ]
     deps = [
       "audio_coding:audio_network_adaptor",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index da6fdccadd..f6188a0425 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -757,6 +757,8 @@ rtc_static_library("audio_network_adaptor") {
     "audio_network_adaptor/frame_length_controller.cc",
     "audio_network_adaptor/frame_length_controller.h",
     "audio_network_adaptor/include/audio_network_adaptor.h",
+    "audio_network_adaptor/smoothing_filter.cc",
+    "audio_network_adaptor/smoothing_filter.h",
   ]
 
   deps = [
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h
index 62a3533a75..0c2388b2c6 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h
@@ -14,8 +14,8 @@
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/common_audio/smoothing_filter.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_unittest.cc
index 823763f091..9bbec26cbe 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_unittest.cc
@@ -10,8 +10,8 @@
 
 #include <utility>
 
-#include "webrtc/common_audio/mocks/mock_smoothing_filter.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
@@ -55,6 +55,7 @@ FecControllerStates CreateFecController(bool initial_fec_enabled) {
   std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
       new NiceMock<MockSmoothingFilter>());
   states.packet_loss_smoothed = mock_smoothing_filter.get();
+  EXPECT_CALL(*states.packet_loss_smoothed, Die());
   using Threshold = FecController::Config::Threshold;
   states.controller.reset(new FecController(
       FecController::Config(
@@ -261,6 +262,7 @@ TEST(FecControllerTest, CheckBehaviorOnSpecialCurves) {
   std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
       new NiceMock<MockSmoothingFilter>());
   states.packet_loss_smoothed = mock_smoothing_filter.get();
+  EXPECT_CALL(*states.packet_loss_smoothed, Die());
   using Threshold = FecController::Config::Threshold;
   states.controller.reset(new FecController(
       FecController::Config(
@@ -291,6 +293,7 @@ TEST(FecControllerDeathTest, InvalidConfig) {
   std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
       new NiceMock<MockSmoothingFilter>());
   states.packet_loss_smoothed = mock_smoothing_filter.get();
+  EXPECT_CALL(*states.packet_loss_smoothed, Die());
   using Threshold = FecController::Config::Threshold;
   EXPECT_DEATH(
       states.controller.reset(new FecController(
diff --git a/webrtc/common_audio/mocks/mock_smoothing_filter.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h
similarity index 60%
rename from webrtc/common_audio/mocks/mock_smoothing_filter.h
rename to webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h
index 6b2991b22f..745ca98ea6 100644
--- a/webrtc/common_audio/mocks/mock_smoothing_filter.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h
@@ -8,20 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
-#define WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
 
-#include "webrtc/common_audio/smoothing_filter.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
 
 class MockSmoothingFilter : public SmoothingFilter {
  public:
+  virtual ~MockSmoothingFilter() { Die(); }
+  MOCK_METHOD0(Die, void());
   MOCK_METHOD1(AddSample, void(float));
   MOCK_CONST_METHOD0(GetAverage, rtc::Optional<float>());
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
diff --git a/webrtc/common_audio/smoothing_filter.cc b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc
similarity index 96%
rename from webrtc/common_audio/smoothing_filter.cc
rename to webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc
index 1cf9580d2a..8a8106918a 100644
--- a/webrtc/common_audio/smoothing_filter.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc
@@ -10,7 +10,7 @@
 
 #include <cmath>
 
-#include "webrtc/common_audio/smoothing_filter.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
 
 namespace webrtc {
 
diff --git a/webrtc/common_audio/smoothing_filter.h b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h
similarity index 83%
rename from webrtc/common_audio/smoothing_filter.h
rename to webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h
index 30d8b7c87d..c4de7b5da5 100644
--- a/webrtc/common_audio/smoothing_filter.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
-#define WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/exp_filter.h"
@@ -46,9 +46,9 @@ class SmoothingFilterImpl final : public SmoothingFilter {
   int64_t last_sample_time_ms_;
   rtc::ExpFilter filter_;
 
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SmoothingFilterImpl);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
diff --git a/webrtc/common_audio/smoothing_filter_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc
similarity index 97%
rename from webrtc/common_audio/smoothing_filter_unittest.cc
rename to webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc
index c80ffb672c..af4e8d957b 100644
--- a/webrtc/common_audio/smoothing_filter_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc
@@ -10,7 +10,7 @@
 
 #include <memory>
 
-#include "webrtc/common_audio/smoothing_filter.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
index b36b16ddbf..2966c29372 100644
--- a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
@@ -9,8 +9,8 @@
  */
 #include <memory>
 
+#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
 #include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
-#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {