diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index d3b1012f70..01a4d7b5f6 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -43,19 +43,19 @@ source_set("audio_processing") {
     "agc/agc_audio_proc_internal.h",
     "agc/agc_manager_direct.cc",
     "agc/agc_manager_direct.h",
-    "agc/analog_agc.c",
-    "agc/analog_agc.h",
     "agc/circular_buffer.cc",
     "agc/circular_buffer.h",
     "agc/common.h",
-    "agc/digital_agc.c",
-    "agc/digital_agc.h",
     "agc/gain_map_internal.h",
     "agc/gmm.cc",
     "agc/gmm.h",
     "agc/histogram.cc",
     "agc/histogram.h",
-    "agc/include/gain_control.h",
+    "agc/legacy/analog_agc.c",
+    "agc/legacy/analog_agc.h",
+    "agc/legacy/digital_agc.c",
+    "agc/legacy/digital_agc.h",
+    "agc/legacy/gain_control.h",
     "agc/noise_gmm_tables.h",
     "agc/pitch_based_vad.cc",
     "agc/pitch_based_vad.h",
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.c b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
similarity index 99%
rename from webrtc/modules/audio_processing/agc/analog_agc.c
rename to webrtc/modules/audio_processing/agc/legacy/analog_agc.c
index 0b50bcb908..734052520b 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
@@ -17,12 +17,13 @@
  *
  */
 
+#include "webrtc/modules/audio_processing/agc/legacy/analog_agc.h"
+
 #include <assert.h>
 #include <stdlib.h>
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
-#include "webrtc/modules/audio_processing/agc/analog_agc.h"
 
 /* The slope of in Q13*/
 static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78};
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.h b/webrtc/modules/audio_processing/agc/legacy/analog_agc.h
similarity index 95%
rename from webrtc/modules/audio_processing/agc/analog_agc.h
rename to webrtc/modules/audio_processing/agc/legacy/analog_agc.h
index 19b324f1ee..a3d02a2a76 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.h
+++ b/webrtc/modules/audio_processing/agc/legacy/analog_agc.h
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-
-#include "webrtc/modules/audio_processing/agc/digital_agc.h"
-#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
-#include "webrtc/typedefs.h"
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
 
 //#define MIC_LEVEL_FEEDBACK
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 
+#include "webrtc/modules/audio_processing/agc/legacy/digital_agc.h"
+#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
+#include "webrtc/typedefs.h"
+
 /* Analog Automatic Gain Control variables:
  * Constant declarations (inner limits inside which no changes are done)
  * In the beginning the range is narrower to widen as soon as the measure
@@ -130,4 +130,4 @@ typedef struct
     int16_t             lowLevelSignal;
 } Agc_t;
 
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.c b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
similarity index 99%
rename from webrtc/modules/audio_processing/agc/digital_agc.c
rename to webrtc/modules/audio_processing/agc/legacy/digital_agc.c
index e09a7fe147..366cd5dfdb 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
@@ -12,7 +12,7 @@
  *
  */
 
-#include "webrtc/modules/audio_processing/agc/digital_agc.h"
+#include "webrtc/modules/audio_processing/agc/legacy/digital_agc.h"
 
 #include <assert.h>
 #include <string.h>
@@ -20,7 +20,7 @@
 #include <stdio.h>
 #endif
 
-#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
+#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
 
 // To generate the gaintable, copy&paste the following lines to a Matlab window:
 // MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.h b/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
similarity index 92%
rename from webrtc/modules/audio_processing/agc/digital_agc.h
rename to webrtc/modules/audio_processing/agc/legacy/digital_agc.h
index 3dcd937516..2d136508da 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.h
+++ b/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
 
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
@@ -75,4 +75,4 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
                                      uint8_t limiterEnable,
                                      int16_t analogTarget);
 
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
diff --git a/webrtc/modules/audio_processing/agc/include/gain_control.h b/webrtc/modules/audio_processing/agc/legacy/gain_control.h
similarity index 97%
rename from webrtc/modules/audio_processing/agc/include/gain_control.h
rename to webrtc/modules/audio_processing/agc/legacy/gain_control.h
index 84f668417d..7eeb78b2f5 100644
--- a/webrtc/modules/audio_processing/agc/include/gain_control.h
+++ b/webrtc/modules/audio_processing/agc/legacy/gain_control.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
 
 #include "webrtc/typedefs.h"
 
@@ -245,4 +245,4 @@ int WebRtcAgc_Init(void *agcInst,
 }
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index a51cf2ea0d..f3eaaa95d4 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -55,19 +55,19 @@
         'agc/agc_audio_proc_internal.h',
         'agc/agc_manager_direct.cc',
         'agc/agc_manager_direct.h',
-        'agc/analog_agc.c',
-        'agc/analog_agc.h',
         'agc/circular_buffer.cc',
         'agc/circular_buffer.h',
         'agc/common.h',
-        'agc/digital_agc.c',
-        'agc/digital_agc.h',
         'agc/gain_map_internal.h',
         'agc/gmm.cc',
         'agc/gmm.h',
         'agc/histogram.cc',
         'agc/histogram.h',
-        'agc/include/gain_control.h',
+        'agc/legacy/analog_agc.c',
+        'agc/legacy/analog_agc.h',
+        'agc/legacy/digital_agc.c',
+        'agc/legacy/digital_agc.h',
+        'agc/legacy/gain_control.h',
         'agc/noise_gmm_tables.h',
         'agc/pitch_based_vad.cc',
         'agc/pitch_based_vad.h',
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index b8fbdc166f..60744a83eb 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -13,7 +13,7 @@
 #include <assert.h>
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
+#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {