diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amr.cc b/webrtc/modules/audio_coding/main/acm2/acm_amr.cc
index 75430f1a8..ab4003abb 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amr.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amr.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_amr.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_amr.h"
 
 #ifdef WEBRTC_CODEC_AMR
 // NOTE! GSM AMR is not included in the open-source package. The following
 // interface file is needed:
 #include "webrtc/modules/audio_coding/main/codecs/amr/interface/amr_interface.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amr.h b/webrtc/modules/audio_coding/main/acm2/acm_amr.h
index c58b5111f..4471e6bca 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amr.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amr.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMR_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct AMR_encinst_t_;
@@ -62,4 +62,4 @@ class ACMAMR : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMR_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc
index 1b82674b1..849353a93 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_amrwb.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_amrwb.h"
 
 #ifdef WEBRTC_CODEC_AMRWB
 // NOTE! GSM AMR-wb is not included in the open-source package. The
 // following interface file is needed:
 #include "webrtc/modules/audio_coding/main/codecs/amrwb/interface/amrwb_interface.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h
index 550bab2d3..e5bd99d9b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMRWB_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMRWB_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct AMRWB_encinst_t_;
@@ -63,4 +63,4 @@ class ACMAMRwb : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMRWB_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_celt.cc b/webrtc/modules/audio_coding/main/acm2/acm_celt.cc
index 6f2c807e1..21fa3a9d0 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_celt.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_celt.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_celt.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_celt.h"
 
 #ifdef WEBRTC_CODEC_CELT
 // NOTE! Celt is not included in the open-source package. Modify this file or
 // your codec API to match the function call and name of used CELT API file.
 #include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_celt.h b/webrtc/modules/audio_coding/main/acm2/acm_celt.h
index b90a4e850..4b40f799e 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_celt.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_celt.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CELT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CELT_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct CELT_encinst_t_;
@@ -47,4 +47,4 @@ class ACMCELT : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CELT_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_cng.cc b/webrtc/modules/audio_coding/main/acm2/acm_cng.cc
index b04fd6ad1..9e658bdad 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_cng.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_cng.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_cng.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_cng.h"
 
 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_cng.h b/webrtc/modules/audio_coding/main/acm2/acm_cng.h
index 2ea4f02db..3816fa2a8 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_cng.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_cng.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CNG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CNG_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct WebRtcCngEncInst;
@@ -53,4 +53,4 @@ class ACMCNG: public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CNG_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
index 08080d1dc..8e14fbbaf 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
@@ -15,22 +15,22 @@
 
 // TODO(tlegrand): Change constant input pointers in all functions to constant
 // references, where appropriate.
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 
 #include <assert.h>
 
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 // Includes needed to create the codecs.
 // G711, PCM mu-law and A-law
-#include "webrtc/modules/audio_coding/main/source/acm_pcma.h"
-#include "webrtc/modules/audio_coding/main/source/acm_pcmu.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_pcma.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_pcmu.h"
 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
 // CNG
 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
-#include "webrtc/modules/audio_coding/main/source/acm_cng.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_cng.h"
 #ifdef WEBRTC_CODEC_ISAC
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
 #endif
@@ -38,66 +38,66 @@
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
 #endif
 #if (defined WEBRTC_CODEC_ISACFX) || (defined WEBRTC_CODEC_ISAC)
-#include "webrtc/modules/audio_coding/main/source/acm_isac.h"
-#include "webrtc/modules/audio_coding/main/source/acm_isac_macros.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_isac.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h"
 #endif
 #ifdef WEBRTC_CODEC_PCM16
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/main/source/acm_pcm16b.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h"
 #endif
 #ifdef WEBRTC_CODEC_ILBC
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
-#include "webrtc/modules/audio_coding/main/source/acm_ilbc.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_ilbc.h"
 #endif
 #ifdef WEBRTC_CODEC_AMR
 #include "webrtc/modules/audio_coding/codecs/amr/include/amr_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_amr.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_amr.h"
 #endif
 #ifdef WEBRTC_CODEC_AMRWB
 #include "webrtc/modules/audio_coding/codecs/amrwb/include/amrwb_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_amrwb.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_amrwb.h"
 #endif
 #ifdef WEBRTC_CODEC_CELT
 #include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_celt.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_celt.h"
 #endif
 #ifdef WEBRTC_CODEC_G722
 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_g722.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g722.h"
 #endif
 #ifdef WEBRTC_CODEC_G722_1
 #include "webrtc/modules/audio_coding/codecs/g7221/include/g7221_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_g7221.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g7221.h"
 #endif
 #ifdef WEBRTC_CODEC_G722_1C
 #include "webrtc/modules/audio_coding/codecs/g7221c/include/g7221c_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_g7221c.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g7221c.h"
 #endif
 #ifdef WEBRTC_CODEC_G729
 #include "webrtc/modules/audio_coding/codecs/g729/include/g729_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_g729.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g729.h"
 #endif
 #ifdef WEBRTC_CODEC_G729_1
 #include "webrtc/modules/audio_coding/codecs/g7291/include/g7291_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_g7291.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g7291.h"
 #endif
 #ifdef WEBRTC_CODEC_GSMFR
 #include "webrtc/modules/audio_coding/codecs/gsmfr/include/gsmfr_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_gsmfr.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h"
 #endif
 #ifdef WEBRTC_CODEC_OPUS
 #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_opus.h"
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
 #include "webrtc/modules/audio_coding/codecs/speex/include/speex_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_speex.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_speex.h"
 #endif
 #ifdef WEBRTC_CODEC_AVT
-#include "webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h"
 #endif
 #ifdef WEBRTC_CODEC_RED
-#include "webrtc/modules/audio_coding/main/source/acm_red.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_red.h"
 #endif
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
index fb5cb9a03..a8a76438c 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
@@ -13,11 +13,11 @@
  * codecs.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
 
 namespace webrtc {
@@ -347,4 +347,4 @@ class ACMCodecDB {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
index b27256a00..39287ea62 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
 
 #include <string.h>
 
@@ -95,4 +95,4 @@ struct WebRtcACMCodecParams {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc
index b6b91029d..ca7e86fd8 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h"
 
 #ifdef WEBRTC_CODEC_AVT
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_receiver.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h
index e16653cdf..4c3154ca9 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DTMF_PLAYOUT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DTMF_PLAYOUT_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 namespace webrtc {
 
@@ -37,4 +37,4 @@ class ACMDTMFPlayout : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DTMF_PLAYOUT_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g722.cc b/webrtc/modules/audio_coding/main/acm2/acm_g722.cc
index 6ba0d7b4a..fe2bd6cb9 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g722.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g722.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_g722.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g722.h"
 
 #ifdef WEBRTC_CODEC_G722
 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g722.h b/webrtc/modules/audio_coding/main/acm2/acm_g722.h
index 21a0fdb2e..34b6c8516 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g722.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g722.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 typedef struct WebRtcG722EncInst G722EncInst;
 typedef struct WebRtcG722DecInst G722DecInst;
@@ -54,4 +54,4 @@ class ACMG722 : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc
index 65b34b08f..0cba71084 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_g7221.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g7221.h"
 
 #ifdef WEBRTC_CODEC_G722_1
 // NOTE! G.722.1 is not included in the open-source package. The following
 // interface file is needed:
 #include "webrtc/modules/audio_coding/main/codecs/g7221/interface/g7221_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 // The API in the header file should match the one below.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221.h b/webrtc/modules/audio_coding/main/acm2/acm_g7221.h
index 2b532db93..4a0bd480d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct G722_1_16_encinst_t_;
@@ -59,4 +59,4 @@ class ACMG722_1 : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc
index b426d1f8b..531008af2 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_g7221c.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g7221c.h"
 
 #ifdef WEBRTC_CODEC_G722_1C
 // NOTE! G.722.1C is not included in the open-source package. The following
 // interface file is needed:
 #include "webrtc/modules/audio_coding/main/codecs/g7221c/interface/g7221c_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 // The API in the header file should match the one below.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h
index d051b28b6..961ed4e17 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221C_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221C_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct G722_1C_24_encinst_t_;
@@ -59,4 +59,4 @@ class ACMG722_1C : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221C_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g729.cc b/webrtc/modules/audio_coding/main/acm2/acm_g729.cc
index a2349ce4a..91dbb43ee 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g729.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g729.cc
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_g729.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g729.h"
 
 #ifdef WEBRTC_CODEC_G729
 // NOTE! G.729 is not included in the open-source package. Modify this file
 // or your codec API to match the function calls and names of used G.729 API
 // file.
 #include "webrtc/modules/audio_coding/main/codecs/g729/interface/g729_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_receiver.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g729.h b/webrtc/modules/audio_coding/main/acm2/acm_g729.h
index 3b35f3b90..f7e762cba 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g729.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g729.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G729_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G729_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct G729_encinst_t_;
@@ -51,4 +51,4 @@ class ACMG729 : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G729_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc
index 1c661c11b..f16eec89b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_g7291.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_g7291.h"
 
 #ifdef WEBRTC_CODEC_G729_1
 // NOTE! G.729.1 is not included in the open-source package. Modify this file
 // or your codec API to match the function calls and names of used G.729.1 API
 // file.
 #include "webrtc/modules/audio_coding/main/codecs/g7291/interface/g7291_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7291.h b/webrtc/modules/audio_coding/main/acm2/acm_g7291.h
index 97601eac1..5a38e59a3 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7291.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7291.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7291_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7291_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct G729_1_inst_t_;
@@ -49,4 +49,4 @@ class ACMG729_1 : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7291_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
index 5e210efd6..4c89b108d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 #include <assert.h>
 #include <string.h>
 
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
index cde2e4284..0129bf38b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_
 
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
@@ -915,4 +915,4 @@ class ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc
index 9fd097c2e..44e6e3d91 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_gsmfr.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h"
 
 #ifdef WEBRTC_CODEC_GSMFR
 // NOTE! GSM-FR is not included in the open-source package. Modify this file
 // or your codec API to match the function calls and names of used GSM-FR API
 // file.
 #include "webrtc/modules/audio_coding/main/codecs/gsmfr/interface/gsmfr_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h
index 935ac444f..51c29eea4 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct GSMFR_encinst_t_;
@@ -47,4 +47,4 @@ class ACMGSMFR : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc
index 204e1e950..14fbbd450 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc
@@ -7,11 +7,11 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/audio_coding/main/source/acm_ilbc.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_ilbc.h"
 
 #ifdef WEBRTC_CODEC_ILBC
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h
index 11e759c6a..e02c789d3 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ILBC_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct iLBC_encinst_t_;
@@ -45,4 +45,4 @@ class ACMILBC : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ILBC_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
index 1c7e8b393..e2de7efb2 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
@@ -7,13 +7,13 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/audio_coding/main/source/acm_isac.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_isac.h"
 
 #include <assert.h>
 
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
@@ -26,7 +26,7 @@
 #endif
 
 #if defined (WEBRTC_CODEC_ISAC) || defined (WEBRTC_CODEC_ISACFX)
-#include "webrtc/modules/audio_coding/main/source/acm_isac_macros.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h"
 #endif
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.h b/webrtc/modules/audio_coding/main/acm2/acm_isac.h
index f4cf1a6ac..2e6657fb4 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 namespace webrtc {
 
@@ -95,4 +95,4 @@ class ACMISAC : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h b/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h
index 646b3cc38..c2a782095 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_MACROS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_MACROS_H_
 
 #include "webrtc/engine_configurations.h"
 
@@ -72,5 +72,5 @@ namespace webrtc {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_MACROS_H_
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_opus.cc b/webrtc/modules/audio_coding/main/acm2/acm_opus.cc
index a7380859c..d627fad8d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_opus.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_opus.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_opus.h"
 
 #ifdef WEBRTC_CODEC_OPUS
 #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_opus.h b/webrtc/modules/audio_coding/main/acm2/acm_opus.h
index 28b08b6fd..caac01093 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_opus.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_opus.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 struct WebRtcOpusEncInst;
 struct WebRtcOpusDecInst;
@@ -47,4 +47,4 @@ class ACMOpus : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc
index 3bc964241..7c5b0bd32 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_pcm16b.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h"
 
 #ifdef WEBRTC_CODEC_PCM16
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h
index a7fff0f52..32490209a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCM16B_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 namespace webrtc {
 
@@ -39,4 +39,4 @@ class ACMPCM16B : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCM16B_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc
index 0d574fe24..cb5ebccfd 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_pcma.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_pcma.h"
 
 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 // Codec interface
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcma.h b/webrtc/modules/audio_coding/main/acm2/acm_pcma.h
index 61386d312..4102e17d9 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcma.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcma.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMA_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMA_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 namespace webrtc {
 
@@ -37,4 +37,4 @@ class ACMPCMA : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMA_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc
index 441e3ddcd..6f479ed21 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_pcmu.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_pcmu.h"
 
 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 // Codec interface.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h
index 832a00d1d..2898df637 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMU_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMU_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 namespace webrtc {
 
@@ -37,4 +37,4 @@ class ACMPCMU : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMU_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index fb3fe3e60..5a36f860b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_receiver.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
 
 #include <stdlib.h>  // malloc
 
@@ -17,9 +17,9 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/source/nack.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/nack.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 344e1c920..5f6d684b0 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RECEIVER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RECEIVER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
 
 #include <vector>
 
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/source/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
@@ -362,4 +362,4 @@ class AcmReceiver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RECEIVER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index ab652cfaa..6fa6743ba 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_receiver.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
 
 #include <algorithm>  // std::min
 
 #include "gtest/gtest.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/test_suite.h"
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_red.cc b/webrtc/modules/audio_coding/main/acm2/acm_red.cc
index 5b5b16f0d..f4a1f6f2a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_red.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_red.cc
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_red.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_red.h"
 
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_red.h b/webrtc/modules/audio_coding/main/acm2/acm_red.h
index c8023db63..ab8d913fa 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_red.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_red.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RED_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RED_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 namespace webrtc {
 
@@ -37,4 +37,4 @@ class ACMRED : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RED_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
index d399cee8e..13eed0ba6 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 
 #include <string.h>
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
index c44fbc47f..8abb2f4f7 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/typedefs.h"
@@ -37,4 +37,4 @@ class ACMResampler {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_speex.cc b/webrtc/modules/audio_coding/main/acm2/acm_speex.cc
index 80dcf5c20..829026549 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_speex.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_speex.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/acm_speex.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_speex.h"
 
 #ifdef WEBRTC_CODEC_SPEEX
 // NOTE! Speex is not included in the open-source package. Modify this file or
 // your codec API to match the function calls and names of used Speex API file.
 #include "webrtc/modules/audio_coding/main/codecs/speex/interface/speex_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #endif
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_speex.h b/webrtc/modules/audio_coding/main/acm2/acm_speex.h
index 68953a8d8..2fac8fd2e 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_speex.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_speex.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SPEEX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SPEEX_H_
 
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
 
 // forward declaration
 struct SPEEX_encinst_t_;
@@ -62,4 +62,4 @@ class ACMSPEEX : public ACMGenericCodec {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SPEEX_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
index 97d5d4628..491160d8b 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -11,8 +11,8 @@
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
index 8b0fbe111..f52625037 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
@@ -7,30 +7,9 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 {
-  'variables': {
-    'audio_coding_dependencies': [
-      'CNG',
-      'G711',
-      'G722',
-      'iLBC',
-      'iSAC',
-      'iSACFix',
-      'PCM16B',
-      'NetEq4',
-      '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
-      '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-    ],
-    'audio_coding_defines': [],
-    'conditions': [
-      ['include_opus==1', {
-        'audio_coding_dependencies': ['webrtc_opus',],
-        'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
-      }],
-    ],
-  },
   'targets': [
     {
-      'target_name': 'audio_coding_module',
+      'target_name': 'acm2',
       'type': 'static_library',
       'defines': [
         '<@(audio_coding_defines)',
@@ -108,83 +87,4 @@
       ],
     },
   ],
-  'conditions': [
-    ['include_tests==1', {
-      'targets': [
-        {
-          'target_name': 'delay_test',
-          'type': 'executable',
-          'dependencies': [
-            'audio_coding_module',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(webrtc_root)/test/test.gyp:test_support_main',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-          ],
-          'sources': [
-             '../test/delay_test.cc',
-             '../test/Channel.cc',
-             '../test/PCMFile.cc',
-          ],
-        }, # delay_test  
-        {
-          # This is handy for testing codecs with different settings. I like to
-          # keep it while we are developing ACM 2. Not sure if we keep it
-          # forever, though I don't have strong reason to remove it.
-          'target_name': 'codec_test',
-          'type': 'executable',
-          'dependencies': [
-            'audio_coding_module',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(webrtc_root)/test/test.gyp:test_support_main',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-          ],
-          'sources': [
-             '../test/codec_test.cc',
-             '../test/Channel.cc',
-             '../test/PCMFile.cc',
-           ],
-        }, # codec_test
-# TODO(turajs): Add this target.        
-#        {
-#          'target_name': 'insert_packet_with_timing',
-#          'type': 'executable',
-#          'dependencies': [
-#            'audio_coding_module',
-#            '<(DEPTH)/testing/gtest.gyp:gtest',
-#            '<(webrtc_root)/test/test.gyp:test_support_main',
-#            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-#            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-#          ],
-#          'sources': [
-#             'acm_receiver_unittest.cc',
-#             '../test/Channel.cc',
-#             '../test/PCMFile.cc',
-#           ],
-#        }, # insert_packet_with_timing
-         {
-          # TODO(turajs): This test will be included in module.gyp when ACM 2 is in 
-          # public repository.
-           'target_name': 'acm2_unittests',
-           'type': 'executable',
-           'defines': [
-             '<@(audio_coding_defines)',
-           ],
-           'dependencies': [
-             'audio_coding_module',
-             '<(DEPTH)/testing/gtest.gyp:gtest',
-             '<(webrtc_root)/test/test.gyp:test_support_main',
-             #'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-           ],
-           'sources': [
-              'nack_unittest.cc',
-              'acm_receiver_unittest.cc',
-              'initial_delay_manager_unittest.cc',
-            ],
-         }, # acm2_unittests
-     ],
-    }],
-  ],
 }
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index f5fac732e..57b79d61d 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
 
 #include <assert.h>
 #include <stdlib.h>
@@ -16,10 +16,10 @@
 
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
-#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index b9c70e9bb..435c7aeab 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
 
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_receiver.h"
-#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
@@ -351,4 +351,4 @@ class AudioCodingModuleImpl : public AudioCodingModule {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
index dffed646c..038b13272 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
index 5c8ae18be..da08f8bd8 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_INITIAL_DELAY_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_INITIAL_DELAY_MANAGER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
 
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
@@ -112,4 +112,4 @@ class InitialDelayManager {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_INITIAL_DELAY_MANAGER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
index 9d96d1770..7e3bda5b5 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
@@ -11,7 +11,7 @@
 #include <cstring>
 
 #include "gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/source/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/acm2/nack.cc b/webrtc/modules/audio_coding/main/acm2/nack.cc
index d4d0c3b8f..e26ad611f 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack.cc
+++ b/webrtc/modules/audio_coding/main/acm2/nack.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/nack.h"
+#include "webrtc/modules/audio_coding/main/acm2/nack.h"
 
 #include <assert.h>  // For assert.
 
diff --git a/webrtc/modules/audio_coding/main/acm2/nack.h b/webrtc/modules/audio_coding/main/acm2/nack.h
index ddafbcdf6..490c03818 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack.h
+++ b/webrtc/modules/audio_coding/main/acm2/nack.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_NACK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_NACK_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
 
 #include <vector>
 #include <map>
@@ -206,4 +206,4 @@ class Nack {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_NACK_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc b/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
index b84211dc3..b047fd6d0 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/source/nack.h"
+#include "webrtc/modules/audio_coding/main/acm2/nack.h"
 
 #include <stdint.h>
 
diff --git a/webrtc/modules/audio_coding/main/source/acm_amr.cc b/webrtc/modules/audio_coding/main/source/acm_amr.cc
index 8e8d6d51f..5590970d6 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amr.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_amr.cc
@@ -49,6 +49,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_AMR
 ACMAMR::ACMAMR(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
@@ -421,4 +423,8 @@ ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const {
 }
 
 #endif
-}
+
+}  // namespace acm1
+
+}  // namespace webrtc
+
diff --git a/webrtc/modules/audio_coding/main/source/acm_amr.h b/webrtc/modules/audio_coding/main/source/acm_amr.h
index 72ed0a22b..19c657246 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amr.h
+++ b/webrtc/modules/audio_coding/main/source/acm_amr.h
@@ -19,7 +19,7 @@ struct AMR_decinst_t_;
 
 namespace webrtc {
 
-enum ACMAMRPackingFormat;
+namespace acm1 {
 
 class ACMAMR : public ACMGenericCodec {
  public:
@@ -80,6 +80,8 @@ class ACMAMR : public ACMGenericCodec {
   ACMAMRPackingFormat decoder_packing_format_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
index fb86a3b42..e2b7635e2 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
@@ -46,6 +46,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_AMRWB
 ACMAMRwb::ACMAMRwb(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
@@ -429,4 +431,6 @@ ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_amrwb.h b/webrtc/modules/audio_coding/main/source/acm_amrwb.h
index 485f1395a..25934187e 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amrwb.h
+++ b/webrtc/modules/audio_coding/main/source/acm_amrwb.h
@@ -19,6 +19,8 @@ struct AMRWB_decinst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMAMRwb : public ACMGenericCodec {
  public:
   explicit ACMAMRwb(int16_t codec_id);
@@ -81,6 +83,8 @@ class ACMAMRwb : public ACMGenericCodec {
   ACMAMRPackingFormat decoder_packing_format_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_celt.cc b/webrtc/modules/audio_coding/main/source/acm_celt.cc
index 31d9e378f..81a034686 100644
--- a/webrtc/modules/audio_coding/main/source/acm_celt.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_celt.cc
@@ -24,6 +24,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_CELT
 
 ACMCELT::ACMCELT(int16_t /* codec_id */)
@@ -332,4 +334,6 @@ void ACMCELT::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_celt.h b/webrtc/modules/audio_coding/main/source/acm_celt.h
index 0bc1afe13..4a4610e0d 100644
--- a/webrtc/modules/audio_coding/main/source/acm_celt.h
+++ b/webrtc/modules/audio_coding/main/source/acm_celt.h
@@ -19,6 +19,8 @@ struct CELT_decinst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMCELT : public ACMGenericCodec {
  public:
   explicit ACMCELT(int16_t codec_id);
@@ -70,6 +72,8 @@ class ACMCELT : public ACMGenericCodec {
   uint16_t dec_channels_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_cng.cc b/webrtc/modules/audio_coding/main/source/acm_cng.cc
index 0c44aa0ab..57c48cd83 100644
--- a/webrtc/modules/audio_coding/main/source/acm_cng.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_cng.cc
@@ -20,6 +20,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 ACMCNG::ACMCNG(int16_t codec_id) {
   encoder_inst_ptr_ = NULL;
   decoder_inst_ptr_ = NULL;
@@ -143,4 +145,6 @@ void ACMCNG::InternalDestructEncoderInst(void* ptr_inst) {
 int16_t ACMCNG::EnableDTX() { return -1; }
 int16_t ACMCNG::DisableDTX() { return -1; }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_cng.h b/webrtc/modules/audio_coding/main/source/acm_cng.h
index 32df58060..728312d55 100644
--- a/webrtc/modules/audio_coding/main/source/acm_cng.h
+++ b/webrtc/modules/audio_coding/main/source/acm_cng.h
@@ -19,6 +19,8 @@ struct WebRtcCngDecInst;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMCNG: public ACMGenericCodec {
  public:
   explicit ACMCNG(int16_t codec_id);
@@ -64,6 +66,8 @@ class ACMCNG: public ACMGenericCodec {
   uint16_t samp_freq_hz_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
index 591f74c72..c3a54d922 100644
--- a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
@@ -101,6 +101,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 // Not yet used payload-types.
 // 83,  82,  81, 80, 79,  78,  77,  76,  75,  74,  73,  72,  71,  70,  69, 68,
 // 67, 66, 65
@@ -949,4 +951,6 @@ bool ACMCodecDB::ValidPayloadType(int payload_type) {
   return true;
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.h b/webrtc/modules/audio_coding/main/source/acm_codec_database.h
index da42a6fa4..7a7054dd1 100644
--- a/webrtc/modules/audio_coding/main/source/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.h
@@ -22,6 +22,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 // TODO(tlegrand): replace class ACMCodecDB with a namespace.
 class ACMCodecDB {
  public:
@@ -327,6 +329,8 @@ class ACMCodecDB {
   static const WebRtcNetEQDecoder neteq_decoders_[kMaxNumCodecs];
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_common_defs.h b/webrtc/modules/audio_coding/main/source/acm_common_defs.h
index b959eeb47..ecc41f8c9 100644
--- a/webrtc/modules/audio_coding/main/source/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/main/source/acm_common_defs.h
@@ -26,6 +26,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 // 60 ms is the maximum block size we support. An extra 20 ms is considered
 // for safety if process() method is not called when it should be, i.e. we
 // accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples.
@@ -104,6 +106,8 @@ struct WebRtcACMAudioBuff {
   uint32_t last_in_timestamp;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc
index 5820bc4ab..edb629876 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc
@@ -14,6 +14,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 ACMDTMFDetection::ACMDTMFDetection() {}
 
 ACMDTMFDetection::~ACMDTMFDetection() {}
@@ -35,4 +37,6 @@ int16_t ACMDTMFDetection::Detect(
   return -1;
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h
index 43a9047a7..74553107a 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h
@@ -17,6 +17,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMDTMFDetection {
  public:
   ACMDTMFDetection();
@@ -33,6 +35,8 @@ class ACMDTMFDetection {
   ACMResampler resampler_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
index 6b91db98e..c8dea7182 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
@@ -18,6 +18,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_AVT
 
 ACMDTMFPlayout::ACMDTMFPlayout(
@@ -164,4 +166,6 @@ void ACMDTMFPlayout::DestructDecoderSafe() {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h
index 11af23495..46175f59e 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMDTMFPlayout: public ACMGenericCodec {
  public:
   explicit ACMDTMFPlayout(int16_t codec_id);
@@ -53,6 +55,8 @@ class ACMDTMFPlayout: public ACMGenericCodec {
   virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g722.cc b/webrtc/modules/audio_coding/main/source/acm_g722.cc
index 1a023db54..5368b35f9 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g722.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g722.cc
@@ -20,6 +20,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_G722
 
 ACMG722::ACMG722(int16_t /* codec_id */)
@@ -351,4 +353,6 @@ void ACMG722::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g722.h b/webrtc/modules/audio_coding/main/source/acm_g722.h
index 8dea5a7a5..cf7ebe1e2 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g722.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g722.h
@@ -18,6 +18,8 @@ typedef struct WebRtcG722DecInst G722DecInst;
 
 namespace webrtc {
 
+namespace acm1 {
+
 // forward declaration
 struct ACMG722EncStr;
 struct ACMG722DecStr;
@@ -75,6 +77,8 @@ class ACMG722 : public ACMGenericCodec {
   G722DecInst* decoder_inst_ptr_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221.cc b/webrtc/modules/audio_coding/main/source/acm_g7221.cc
index f784b6226..c9074ac7c 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221.cc
@@ -86,6 +86,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_G722_1
 
 ACMG722_1::ACMG722_1(int16_t /* codec_id */)
@@ -493,4 +495,6 @@ void ACMG722_1::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221.h b/webrtc/modules/audio_coding/main/source/acm_g7221.h
index 4e35476d4..8ea66742c 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221.h
@@ -24,6 +24,8 @@ struct G722_1_Inst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMG722_1: public ACMGenericCodec {
  public:
   explicit ACMG722_1(int16_t codec_id);
@@ -77,6 +79,8 @@ class ACMG722_1: public ACMGenericCodec {
   G722_1_32_decinst_t_* decoder_inst32_ptr_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
index a0d94836a..91071e9b4 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
@@ -87,6 +87,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_G722_1C
 
 ACMG722_1C::ACMG722_1C(int16_t /* codec_id */)
@@ -503,4 +505,6 @@ void ACMG722_1C::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221c.h b/webrtc/modules/audio_coding/main/source/acm_g7221c.h
index 1b4e75649..d8875aa2f 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221c.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221c.h
@@ -24,6 +24,8 @@ struct G722_1_Inst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMG722_1C : public ACMGenericCodec {
  public:
   explicit ACMG722_1C(int16_t codec_id);
@@ -85,6 +87,8 @@ class ACMG722_1C : public ACMGenericCodec {
   G722_1C_48_decinst_t_* decoder_inst48_ptr_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc;
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g729.cc b/webrtc/modules/audio_coding/main/source/acm_g729.cc
index 67611cbdc..5b75ab948 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g729.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g729.cc
@@ -25,6 +25,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_G729
 
 ACMG729::ACMG729(int16_t /* codec_id */)
@@ -359,4 +361,6 @@ void ACMG729::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g729.h b/webrtc/modules/audio_coding/main/source/acm_g729.h
index d50aa5f3f..5cfff63b6 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g729.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g729.h
@@ -19,6 +19,8 @@ struct G729_decinst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMG729 : public ACMGenericCodec {
  public:
   explicit ACMG729(int16_t codec_id);
@@ -67,6 +69,8 @@ class ACMG729 : public ACMGenericCodec {
 
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7291.cc b/webrtc/modules/audio_coding/main/source/acm_g7291.cc
index da473ca84..fd241b393 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7291.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7291.cc
@@ -24,6 +24,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_G729_1
 
 ACMG729_1::ACMG729_1(int16_t /* codec_id */)
@@ -342,4 +344,6 @@ int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7291.h b/webrtc/modules/audio_coding/main/source/acm_g7291.h
index 433b2fdf8..bac7faf83 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7291.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g7291.h
@@ -19,6 +19,8 @@ struct G729_1_inst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMG729_1 : public ACMGenericCodec {
  public:
   explicit ACMG729_1(int16_t codec_id);
@@ -63,6 +65,8 @@ class ACMG729_1 : public ACMGenericCodec {
   int16_t flag_g729_mode_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
index 94aeb4837..52f51146b 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
@@ -22,6 +22,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 // Enum for CNG
 enum {
   kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER,
@@ -1251,4 +1253,6 @@ int16_t ACMGenericCodec::REDPayloadISAC(const int32_t /* isac_rate */,
 
 bool ACMGenericCodec::IsTrueStereoCodec() { return false; }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
index 9ba8d08fd..3951a94d5 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
@@ -27,6 +27,9 @@ namespace webrtc {
 
 // forward declaration
 struct CodecInst;
+
+namespace acm1 {
+
 class ACMNetEQ;
 
 class ACMGenericCodec {
@@ -1213,6 +1216,8 @@ class ACMGenericCodec {
   uint32_t unique_id_;
 };
 
-}  // namespace webrt
+}  // namespace acm1
+
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
index 22bbbd8f8..9fa041064 100644
--- a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
@@ -24,6 +24,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_GSMFR
 
 ACMGSMFR::ACMGSMFR(int16_t /* codec_id */)
@@ -260,4 +262,6 @@ void ACMGSMFR::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_gsmfr.h b/webrtc/modules/audio_coding/main/source/acm_gsmfr.h
index 61f576891..aa499734a 100644
--- a/webrtc/modules/audio_coding/main/source/acm_gsmfr.h
+++ b/webrtc/modules/audio_coding/main/source/acm_gsmfr.h
@@ -19,6 +19,8 @@ struct GSMFR_decinst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMGSMFR : public ACMGenericCodec {
  public:
   explicit ACMGSMFR(int16_t codec_id);
@@ -62,6 +64,8 @@ class ACMGSMFR : public ACMGenericCodec {
   GSMFR_decinst_t_* decoder_inst_ptr_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
index a2a294ef5..b47e75090 100644
--- a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_ILBC
 
 ACMILBC::ACMILBC(int16_t /* codec_id */)
@@ -252,4 +254,6 @@ int16_t ACMILBC::SetBitRateSafe(const int32_t rate) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_ilbc.h b/webrtc/modules/audio_coding/main/source/acm_ilbc.h
index 50b6df9a9..bd2495fe3 100644
--- a/webrtc/modules/audio_coding/main/source/acm_ilbc.h
+++ b/webrtc/modules/audio_coding/main/source/acm_ilbc.h
@@ -19,6 +19,8 @@ struct iLBC_decinst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMILBC : public ACMGenericCodec {
  public:
   explicit ACMILBC(int16_t codec_id);
@@ -62,6 +64,8 @@ class ACMILBC : public ACMGenericCodec {
   iLBC_decinst_t_* decoder_inst_ptr_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_isac.cc b/webrtc/modules/audio_coding/main/source/acm_isac.cc
index e22d3f61b..b9316d6d9 100644
--- a/webrtc/modules/audio_coding/main/source/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_isac.cc
@@ -28,6 +28,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 // we need this otherwise we cannot use forward declaration
 // in the header file
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
@@ -896,4 +898,6 @@ int16_t ACMISAC::REDPayloadISAC(const int32_t isac_rate,
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_isac.h b/webrtc/modules/audio_coding/main/source/acm_isac.h
index 9588723cb..20b6c5391 100644
--- a/webrtc/modules/audio_coding/main/source/acm_isac.h
+++ b/webrtc/modules/audio_coding/main/source/acm_isac.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 struct ACMISACInst;
 
 enum IsacCodingMode {
@@ -129,6 +131,8 @@ class ACMISAC : public ACMGenericCodec {
   WebRtcACMCodecParams decoder_params_32khz_;
 };
 
-}  // namespace
+}  // namespace acm1
+
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_isac_macros.h b/webrtc/modules/audio_coding/main/source/acm_isac_macros.h
index 6ae4526f5..01e1e44b3 100644
--- a/webrtc/modules/audio_coding/main/source/acm_isac_macros.h
+++ b/webrtc/modules/audio_coding/main/source/acm_isac_macros.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifdef WEBRTC_CODEC_ISAC
 #define ACM_ISAC_CREATE            WebRtcIsac_Create
 #define ACM_ISAC_FREE              WebRtcIsac_Free
@@ -67,7 +69,9 @@ namespace webrtc {
 #define ACM_ISAC_GETDECSAMPRATE    ACMISACFixGetDecSampRate  // local Impl
 #endif
 
-}  // namespace
+}  // namespace acm1
+
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
 
diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq.cc b/webrtc/modules/audio_coding/main/source/acm_neteq.cc
index 5418d1802..2ade7bf57 100644
--- a/webrtc/modules/audio_coding/main/source/acm_neteq.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_neteq.cc
@@ -26,6 +26,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #define RTP_HEADER_SIZE 12
 #define NETEQ_INIT_FREQ 8000
 #define NETEQ_INIT_FREQ_KHZ (NETEQ_INIT_FREQ/1000)
@@ -1140,4 +1142,6 @@ bool ACMNetEQ::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const {
   return true;
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq.h b/webrtc/modules/audio_coding/main/source/acm_neteq.h
index 511968b28..e52ddc795 100644
--- a/webrtc/modules/audio_coding/main/source/acm_neteq.h
+++ b/webrtc/modules/audio_coding/main/source/acm_neteq.h
@@ -12,8 +12,6 @@
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
 
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
 #include "webrtc/modules/interface/module_common_types.h"
@@ -25,6 +23,8 @@ class CriticalSectionWrapper;
 class RWLockWrapper;
 struct CodecInst;
 
+namespace acm1 {
+
 #define MAX_NUM_SLAVE_NETEQ 1
 
 class ACMNetEQ {
@@ -392,6 +392,8 @@ class ACMNetEQ {
   int maximum_delay_ms_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc b/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc
index aef0acdde..8b973ba23 100644
--- a/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc
@@ -24,6 +24,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class AcmNetEqTest : public ::testing::Test {
  protected:
   static const size_t kMaxPayloadLen = 5760;  // 60 ms, 48 kHz, 16 bit samples.
@@ -146,4 +148,6 @@ TEST_F(AcmNetEqTest, TestZeroLengthWaitingTimesVector) {
   EXPECT_EQ(-1, stats.medianWaitingTimeMs);
 }
 
-}  // namespace
+}  // namespace acm1
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.cc b/webrtc/modules/audio_coding/main/source/acm_opus.cc
index 8ea5d51d5..3a619d04e 100644
--- a/webrtc/modules/audio_coding/main/source/acm_opus.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_opus.cc
@@ -23,6 +23,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_OPUS
 
 ACMOpus::ACMOpus(int16_t /* codec_id */)
@@ -312,4 +314,6 @@ void ACMOpus::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
 
 #endif  // WEBRTC_CODEC_OPUS
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.h b/webrtc/modules/audio_coding/main/source/acm_opus.h
index fa188a1cb..1e586ff41 100644
--- a/webrtc/modules/audio_coding/main/source/acm_opus.h
+++ b/webrtc/modules/audio_coding/main/source/acm_opus.h
@@ -19,6 +19,8 @@ struct WebRtcOpusDecInst;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMOpus : public ACMGenericCodec {
  public:
   explicit ACMOpus(int16_t codec_id);
@@ -69,6 +71,8 @@ class ACMOpus : public ACMGenericCodec {
   int channels_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
index 91cb9e03d..b0032b860 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
@@ -23,6 +23,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_PCM16
 
 ACMPCM16B::ACMPCM16B(int16_t /* codec_id */) {
@@ -244,4 +246,6 @@ void ACMPCM16B::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
 }
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcm16b.h b/webrtc/modules/audio_coding/main/source/acm_pcm16b.h
index 38de34376..a97589b57 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcm16b.h
+++ b/webrtc/modules/audio_coding/main/source/acm_pcm16b.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMPCM16B : public ACMGenericCodec {
  public:
   explicit ACMPCM16B(int16_t codec_id);
@@ -58,6 +60,8 @@ class ACMPCM16B : public ACMGenericCodec {
   int32_t sampling_freq_hz_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcma.cc b/webrtc/modules/audio_coding/main/source/acm_pcma.cc
index 83c124922..c64641771 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcma.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcma.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 ACMPCMA::ACMPCMA(int16_t codec_id) {
   codec_id_ = codec_id;
 }
@@ -127,4 +129,6 @@ void ACMPCMA::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
   }
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcma.h b/webrtc/modules/audio_coding/main/source/acm_pcma.h
index 2fc4ea4fe..cb506eaa6 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcma.h
+++ b/webrtc/modules/audio_coding/main/source/acm_pcma.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMPCMA : public ACMGenericCodec {
  public:
   explicit ACMPCMA(int16_t codec_id);
@@ -56,6 +58,8 @@ class ACMPCMA : public ACMGenericCodec {
                                  int32_t* payload_length) OVERRIDE;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
index 61a64ace9..5b6a4575f 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 ACMPCMU::ACMPCMU(int16_t codec_id) {
   codec_id_ = codec_id;
 }
@@ -129,4 +131,6 @@ void ACMPCMU::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
   }
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcmu.h b/webrtc/modules/audio_coding/main/source/acm_pcmu.h
index 309d31870..ea401d59c 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcmu.h
+++ b/webrtc/modules/audio_coding/main/source/acm_pcmu.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMPCMU : public ACMGenericCodec {
  public:
   explicit ACMPCMU(int16_t codec_id);
@@ -56,6 +58,8 @@ class ACMPCMU : public ACMGenericCodec {
                                  int32_t* payload_length) OVERRIDE;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_red.cc b/webrtc/modules/audio_coding/main/source/acm_red.cc
index 6e7ae9fe7..bc44c7231 100644
--- a/webrtc/modules/audio_coding/main/source/acm_red.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_red.cc
@@ -18,6 +18,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 ACMRED::ACMRED(int16_t codec_id) {
   codec_id_ = codec_id;
 }
@@ -101,4 +103,6 @@ void ACMRED::DestructDecoderSafe() {
   return;
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_red.h b/webrtc/modules/audio_coding/main/source/acm_red.h
index c7bad4163..ede18b521 100644
--- a/webrtc/modules/audio_coding/main/source/acm_red.h
+++ b/webrtc/modules/audio_coding/main/source/acm_red.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMRED : public ACMGenericCodec {
  public:
   explicit ACMRED(int16_t codec_id);
@@ -53,6 +55,8 @@ class ACMRED : public ACMGenericCodec {
   virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_resampler.cc b/webrtc/modules/audio_coding/main/source/acm_resampler.cc
index 034dbe550..50ddab1d8 100644
--- a/webrtc/modules/audio_coding/main/source/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_resampler.cc
@@ -17,6 +17,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 ACMResampler::ACMResampler() {
 }
 
@@ -56,4 +58,6 @@ int16_t ACMResampler::Resample10Msec(const int16_t* in_audio,
   return out_length / num_audio_channels;
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_resampler.h b/webrtc/modules/audio_coding/main/source/acm_resampler.h
index c23abb882..b50e722c4 100644
--- a/webrtc/modules/audio_coding/main/source/acm_resampler.h
+++ b/webrtc/modules/audio_coding/main/source/acm_resampler.h
@@ -16,6 +16,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMResampler {
  public:
   ACMResampler();
@@ -31,6 +33,8 @@ class ACMResampler {
   PushResampler resampler_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_speex.cc b/webrtc/modules/audio_coding/main/source/acm_speex.cc
index ce205266a..575269347 100644
--- a/webrtc/modules/audio_coding/main/source/acm_speex.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_speex.cc
@@ -25,6 +25,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 #ifndef WEBRTC_CODEC_SPEEX
 ACMSPEEX::ACMSPEEX(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
@@ -464,4 +466,6 @@ int16_t ACMSPEEX::SetComplMode(int16_t mode) {
 
 #endif
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_speex.h b/webrtc/modules/audio_coding/main/source/acm_speex.h
index 0f62ea34a..762aea8d9 100644
--- a/webrtc/modules/audio_coding/main/source/acm_speex.h
+++ b/webrtc/modules/audio_coding/main/source/acm_speex.h
@@ -19,6 +19,8 @@ struct SPEEX_decinst_t_;
 
 namespace webrtc {
 
+namespace acm1 {
+
 class ACMSPEEX : public ACMGenericCodec {
  public:
   explicit ACMSPEEX(int16_t codec_id);
@@ -77,6 +79,8 @@ class ACMSPEEX : public ACMGenericCodec {
   uint16_t samples_in_20ms_audio_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
index 4ac40b77d..9461a1f18 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
@@ -21,31 +21,31 @@ namespace webrtc {
 
 // Create module
 AudioCodingModule* AudioCodingModule::Create(const int32_t id) {
-  return new AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
+  return new acm1::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
 }
 
 // Used for testing by inserting a simulated clock. ACM will not destroy the
 // injected |clock| the client has to take care of that.
 AudioCodingModule* AudioCodingModule::Create(const int32_t id,
                                              Clock* clock) {
-  return new AudioCodingModuleImpl(id, clock);
+  return new acm1::AudioCodingModuleImpl(id, clock);
 }
 
 // Destroy module
 void AudioCodingModule::Destroy(AudioCodingModule* module) {
-  delete static_cast<AudioCodingModuleImpl*>(module);
+  delete static_cast<acm1::AudioCodingModuleImpl*>(module);
 }
 
 // Get number of supported codecs
 uint8_t AudioCodingModule::NumberOfCodecs() {
-  return static_cast<uint8_t>(ACMCodecDB::kNumCodecs);
+  return static_cast<uint8_t>(acm1::ACMCodecDB::kNumCodecs);
 }
 
 // Get supported codec param with id
 int32_t AudioCodingModule::Codec(uint8_t list_id,
                                  CodecInst* codec) {
   // Get the codec settings for the codec with the given list ID
-  return ACMCodecDB::Codec(list_id, codec);
+  return acm1::ACMCodecDB::Codec(list_id, codec);
 }
 
 // Get supported codec Param with name, frequency and number of channels.
@@ -55,7 +55,8 @@ int32_t AudioCodingModule::Codec(const char* payload_name,
   int codec_id;
 
   // Get the id of the codec from the database.
-  codec_id = ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
+  codec_id = acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz,
+                                       channels);
   if (codec_id < 0) {
     // We couldn't find a matching codec, set the parameters to unacceptable
     // values and return.
@@ -68,7 +69,7 @@ int32_t AudioCodingModule::Codec(const char* payload_name,
   }
 
   // Get default codec settings.
-  ACMCodecDB::Codec(codec_id, codec);
+  acm1::ACMCodecDB::Codec(codec_id, codec);
 
   // Keep the number of channels from the function call. For most codecs it
   // will be the same value as in default codec settings, but not for all.
@@ -80,14 +81,14 @@ int32_t AudioCodingModule::Codec(const char* payload_name,
 // Get supported codec Index with name, frequency and number of channels.
 int32_t AudioCodingModule::Codec(const char* payload_name,
                                  int sampling_freq_hz, int channels) {
-  return ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
+  return acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
 }
 
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   int mirror_id;
 
-  int codec_number = ACMCodecDB::CodecNumber(&codec, &mirror_id);
+  int codec_number = acm1::ACMCodecDB::CodecNumber(&codec, &mirror_id);
 
   if (codec_number < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
@@ -99,8 +100,8 @@ bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
 }
 
 AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
-  return new AudioCodingModuleImpl(static_cast<int32_t>(id),
-                                   Clock::GetRealTimeClock());
+  return new acm1::AudioCodingModuleImpl(static_cast<int32_t>(id),
+                                         Clock::GetRealTimeClock());
 }
 
 AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index b136d84ca..93b21e68d 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -32,6 +32,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 enum {
   kACMToneEnd = 999
 };
@@ -3115,4 +3117,6 @@ void AudioCodingModuleImpl::DisableNack() {
   nack_enabled_ = false;
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
index f58e3e5ed..64afe4f8e 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -15,6 +15,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
 #include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
@@ -22,11 +23,16 @@
 
 namespace webrtc {
 
-class ACMDTMFDetection;
-class ACMGenericCodec;
 class CriticalSectionWrapper;
 class RWLockWrapper;
 class Clock;
+
+namespace acm1 {
+
+struct WebRtcACMAudioBuff;
+struct WebRtcACMCodecParams;
+class ACMDTMFDetection;
+class ACMGenericCodec;
 class Nack;
 
 class AudioCodingModuleImpl : public AudioCodingModule {
@@ -88,8 +94,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
 
   // Register a transport callback which will be
   // called to deliver the encoded buffers.
-  int32_t RegisterTransportCallback(
-      AudioPacketizationCallback* transport);
+  int32_t RegisterTransportCallback(AudioPacketizationCallback* transport);
 
   // Used by the module to deliver messages to the codec module/application
   // AVT(DTMF).
@@ -125,8 +130,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
                  bool enable_vad = false,
                  ACMVADMode mode = VADNormal);
 
-  int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
-              ACMVADMode* mode) const;
+  int32_t VAD(bool* dtx_enabled, bool* vad_enabled, ACMVADMode* mode) const;
 
   int32_t RegisterVADCallback(ACMVADCallback* vad_callback);
 
@@ -454,6 +458,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   bool nack_enabled_;
 };
 
+}  // namespace acm1
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/webrtc/modules/audio_coding/main/source/nack.cc b/webrtc/modules/audio_coding/main/source/nack.cc
index ec6cb3d12..4ca260ddc 100644
--- a/webrtc/modules/audio_coding/main/source/nack.cc
+++ b/webrtc/modules/audio_coding/main/source/nack.cc
@@ -19,6 +19,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 namespace {
 
 const int kDefaultSampleRateKhz = 48;
@@ -222,4 +224,6 @@ std::vector<uint16_t> Nack::GetNackList(int round_trip_time_ms) const {
   return sequence_numbers;
 }
 
-}  // webrtc
+}  // namespace acm1
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/nack.h b/webrtc/modules/audio_coding/main/source/nack.h
index e047c2824..9cea15d1a 100644
--- a/webrtc/modules/audio_coding/main/source/nack.h
+++ b/webrtc/modules/audio_coding/main/source/nack.h
@@ -49,6 +49,8 @@
 //
 namespace webrtc {
 
+namespace acm1 {
+
 class Nack {
  public:
   // A limit for the size of the NACK list.
@@ -204,6 +206,8 @@ class Nack {
   size_t max_nack_list_size_;
 };
 
-}  // webrtc
+}  // namespace acm1
+
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_NACK_H_
diff --git a/webrtc/modules/audio_coding/main/source/nack_unittest.cc b/webrtc/modules/audio_coding/main/source/nack_unittest.cc
index ba92f0e76..811aca4fc 100644
--- a/webrtc/modules/audio_coding/main/source/nack_unittest.cc
+++ b/webrtc/modules/audio_coding/main/source/nack_unittest.cc
@@ -13,6 +13,7 @@
 #include <stdint.h>
 
 #include <algorithm>
+#include <vector>
 
 #include "gtest/gtest.h"
 #include "webrtc/typedefs.h"
@@ -21,6 +22,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 namespace {
 
 const int kNackThreshold = 3;
@@ -479,4 +482,6 @@ TEST(NackTest, RoudTripTimeIsApplied) {
   EXPECT_EQ(5, nack_list[1]);
 }
 
-}  // webrtc
+}  // namespace acm1
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/main/test/opus_test.h
index ca0da9ee1..49b98ea86 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.h
+++ b/webrtc/modules/audio_coding/main/test/opus_test.h
@@ -29,6 +29,7 @@ class OpusTest : public ACMTest {
   ~OpusTest();
 
   void Perform();
+
  private:
   void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
            int percent_loss = 0);
@@ -44,7 +45,7 @@ class OpusTest : public ACMTest {
   int counter_;
   uint8_t payload_type_;
   int rtp_timestamp_;
-  ACMResampler resampler_;
+  acm1::ACMResampler resampler_;
   WebRtcOpusEncInst* opus_mono_encoder_;
   WebRtcOpusEncInst* opus_stereo_encoder_;
   WebRtcOpusDecInst* opus_mono_decoder_;