diff --git a/.gn b/.gn
index 3e0fcf275e..77394c6440 100644
--- a/.gn
+++ b/.gn
@@ -42,6 +42,9 @@ check_targets = [
   "//webrtc/modules/remote_bitrate_estimator/*",
   "//webrtc/stats:rtc_stats",
   "//webrtc/voice_engine",
+  "//webrtc/voice_engine:audio_coder",
+  "//webrtc/voice_engine:file_player",
+  "//webrtc/voice_engine:file_recorder",
   "//webrtc/voice_engine:level_indicator",
 ]
 
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index bbb5744422..37b5b12ff3 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -231,8 +231,6 @@ if (rtc_include_tests) {
     "//resources/synthetic-trace.rx",
     "//resources/tmobile-downlink.rx",
     "//resources/tmobile-uplink.rx",
-    "//resources/utility/encapsulated_pcm16b_8khz.wav",
-    "//resources/utility/encapsulated_pcmu_8khz.wav",
     "//resources/verizon3g-downlink.rx",
     "//resources/verizon3g-uplink.rx",
     "//resources/verizon4g-downlink.rx",
@@ -483,7 +481,6 @@ if (rtc_include_tests) {
       "rtp_rtcp/test/testAPI/test_api_audio.cc",
       "rtp_rtcp/test/testAPI/test_api_rtcp.cc",
       "rtp_rtcp/test/testAPI/test_api_video.cc",
-      "utility/source/file_player_unittests.cc",
       "utility/source/process_thread_impl_unittest.cc",
       "video_coding/codecs/test/packet_manipulator_unittest.cc",
       "video_coding/codecs/test/stats_unittest.cc",
diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn
index edd428fe09..48cbf793ed 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -11,15 +11,9 @@ import("../../build/webrtc.gni")
 rtc_static_library("utility") {
   sources = [
     "include/audio_frame_operations.h",
-    "include/file_player.h",
-    "include/file_recorder.h",
     "include/helpers_android.h",
     "include/jvm_android.h",
     "include/process_thread.h",
-    "source/coder.cc",
-    "source/coder.h",
-    "source/file_player.cc",
-    "source/file_recorder.cc",
     "source/helpers_android.cc",
     "source/jvm_android.cc",
     "source/process_thread_impl.cc",
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 6413c1842a..0591999def 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -8,6 +8,64 @@
 
 import("../build/webrtc.gni")
 
+rtc_static_library("audio_coder") {
+  sources = [
+    "coder.cc",
+    "coder.h",
+  ]
+  deps = [
+    "..:webrtc_common",
+    "../modules/audio_coding",
+    "../modules/audio_coding:audio_format_conversion",
+    "../modules/audio_coding:builtin_audio_decoder_factory",
+    "../modules/audio_coding:rent_a_codec",
+  ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+rtc_static_library("file_player") {
+  sources = [
+    "file_player.cc",
+    "file_player.h",
+  ]
+  deps = [
+    ":audio_coder",
+    "..:webrtc_common",
+    "../common_audio",
+    "../modules/media_file",
+    "../system_wrappers",
+  ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+rtc_static_library("file_recorder") {
+  sources = [
+    "file_recorder.cc",
+    "file_recorder.h",
+  ]
+  deps = [
+    ":audio_coder",
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../common_audio",
+    "../modules/media_file:media_file",
+    "../system_wrappers",
+  ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
 rtc_static_library("voice_engine") {
   sources = [
     "channel.cc",
@@ -85,6 +143,8 @@ rtc_static_library("voice_engine") {
     "../modules/audio_coding",
   ]
   deps = [
+    ":file_player",
+    ":file_recorder",
     ":level_indicator",
     "..:webrtc_common",
     "../api:audio_mixer_api",
@@ -191,6 +251,7 @@ if (rtc_include_tests) {
       ":voice_engine",
       "//testing/gmock",
       "//testing/gtest",
+      "//third_party/gflags",
       "//webrtc/common_audio",
       "//webrtc/modules/audio_coding",
       "//webrtc/modules/audio_conference_mixer",
@@ -210,6 +271,7 @@ if (rtc_include_tests) {
 
     sources = [
       "channel_unittest.cc",
+      "file_player_unittests.cc",
       "test/channel_transport/udp_socket_manager_unittest.cc",
       "test/channel_transport/udp_socket_wrapper_unittest.cc",
       "test/channel_transport/udp_transport_unittest.cc",
@@ -223,6 +285,11 @@ if (rtc_include_tests) {
       "voice_engine_fixture.h",
     ]
 
+    data = [
+      "//resources/utility/encapsulated_pcm16b_8khz.wav",
+      "//resources/utility/encapsulated_pcmu_8khz.wav",
+    ]
+
     if (is_win) {
       defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
 
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 01907d269b..11e109e3c8 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -28,8 +28,8 @@
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/utility/include/file_player.h"
-#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/voice_engine/file_player.h"
+#include "webrtc/voice_engine/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_network.h"
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/voice_engine/coder.cc
similarity index 98%
rename from webrtc/modules/utility/source/coder.cc
rename to webrtc/voice_engine/coder.cc
index 71f969097f..82eb2487b6 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/voice_engine/coder.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/voice_engine/coder.h"
+
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/source/coder.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/voice_engine/coder.h
similarity index 93%
rename from webrtc/modules/utility/source/coder.h
rename to webrtc/voice_engine/coder.h
index 4855a00465..5e16b0af66 100644
--- a/webrtc/modules/utility/source/coder.h
+++ b/webrtc/voice_engine/coder.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+#ifndef WEBRTC_VOICE_ENGINE_CODER_H_
+#define WEBRTC_VOICE_ENGINE_CODER_H_
 
 #include <memory>
 
@@ -65,4 +65,4 @@ class AudioCoder : public AudioPacketizationCallback {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+#endif  // WEBRTC_VOICE_ENGINE_CODER_H_
diff --git a/webrtc/modules/utility/source/file_player.cc b/webrtc/voice_engine/file_player.cc
similarity index 99%
rename from webrtc/modules/utility/source/file_player.cc
rename to webrtc/voice_engine/file_player.cc
index fa5bc3d2e4..b581d5235b 100644
--- a/webrtc/modules/utility/source/file_player.cc
+++ b/webrtc/voice_engine/file_player.cc
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/voice_engine/file_player.h"
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
-#include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/voice_engine/coder.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/voice_engine/file_player.h
similarity index 94%
rename from webrtc/modules/utility/include/file_player.h
rename to webrtc/voice_engine/file_player.h
index 1adbb9d70e..956016f07f 100644
--- a/webrtc/modules/utility/include/file_player.h
+++ b/webrtc/voice_engine/file_player.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
+#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
 
 #include <memory>
 
@@ -77,4 +77,4 @@ class FilePlayer {
   virtual int32_t SetAudioScaling(float scaleFactor) = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#endif  // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/voice_engine/file_player_unittests.cc
similarity index 98%
rename from webrtc/modules/utility/source/file_player_unittests.cc
rename to webrtc/voice_engine/file_player_unittests.cc
index cc7865d150..d41f4044b4 100644
--- a/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/webrtc/voice_engine/file_player_unittests.cc
@@ -10,7 +10,7 @@
 
 // Unit tests for FilePlayer.
 
-#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/voice_engine/file_player.h"
 
 #include <stdio.h>
 
diff --git a/webrtc/modules/utility/source/file_recorder.cc b/webrtc/voice_engine/file_recorder.cc
similarity index 98%
rename from webrtc/modules/utility/source/file_recorder.cc
rename to webrtc/voice_engine/file_recorder.cc
index 34bd3945fa..9a0edb057b 100644
--- a/webrtc/modules/utility/source/file_recorder.cc
+++ b/webrtc/voice_engine/file_recorder.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/voice_engine/file_recorder.h"
 
 #include <list>
 
@@ -18,10 +18,10 @@
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
-#include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/voice_engine/coder.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/utility/include/file_recorder.h b/webrtc/voice_engine/file_recorder.h
similarity index 91%
rename from webrtc/modules/utility/include/file_recorder.h
rename to webrtc/voice_engine/file_recorder.h
index 6d4d321354..c4195d0eba 100644
--- a/webrtc/modules/utility/include/file_recorder.h
+++ b/webrtc/voice_engine/file_recorder.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
+#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
 
 #include <memory>
 
@@ -54,4 +54,4 @@ class FileRecorder {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#endif  // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index cc8dddfbfc..5e3e2e708e 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -18,7 +18,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/voice_engine/file_recorder.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 3147c1036f..13ddaa5492 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -18,8 +18,8 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/typing_detection.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/file_player.h"
-#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/voice_engine/file_player.h"
+#include "webrtc/voice_engine/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"