Prepare to compile ACM1 and ACM2.

ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2206004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amr.cc b/webrtc/modules/audio_coding/main/acm2/acm_amr.cc
index 75430f1..ab4003a 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 c58b511..4471e6b 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 @@
 
 }  // 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 1b82674..849353a 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 550bab2..e5bd99d 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 @@
 
 }  // 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 6f2c807..21fa3a9 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 b90a4e8..4b40f79 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 @@
 
 }  // 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 b04fd6a..9e658bd 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 2ea4f02..3816fa2 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 @@
 
 }  // 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 08080d1..8e14fbb 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 fb5cb9a..a8a7643 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 @@
 
 }  // 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 b27256a..39287ea 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 @@
 
 }  // 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 b6b9102..ca7e86f 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 e16653c..4c3154c 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 @@
 
 }  // 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 6ba0d7b..fe2bd6c 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 21a0fdb..34b6c85 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 @@
 
 }  // 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 65b34b0..0cba710 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 2b532db9..4a0bd48 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 @@
 
 }  // 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 b426d1f..531008a 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 d051b28..961ed4e 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 @@
 
 }  // 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 a2349ce..91dbb43 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 3b35f3b..f7e762c 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 @@
 
 }  // 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 1c661c1..f16eec8 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 97601ea..5a38e59 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 @@
 
 }  // 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 5e210ef..4c89b10 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 cde2e42..0129bf3 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 @@
 
 }  // 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 9fd097c..44e6e3d 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 935ac44..51c29ee 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 @@
 
 }  // 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 204e1e9..14fbbd4 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 11e759c..e02c789 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 @@
 
 }  // 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 1c7e8b3..e2de7ef 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 f4cf1a6..2e6657f 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 @@
 
 }  // 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 646b3cc..c2a7820 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
 
-#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 a738085..d627fad 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 28b08b6..caac010 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 @@
 
 }  // 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 3bc9642..7c5b0bd 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 a7fff0f..3249020 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 @@
 
 }  // 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 0d574fe..cb5ebcc 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 61386d3..4102e17 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 @@
 
 }  // 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 441e3dd..6f479ed 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 832a00d..2898df6 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 @@
 
 }  // 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 fb3fe3e..5a36f86 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 344e1c9..5f6d684 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 @@
 
 }  // 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 ab652cf..6fa6743 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 5b5b16f..f4a1f6f 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 c8023db..ab8d913 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 @@
 
 }  // 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 d399cee..13eed0b 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 c44fbc4..8abb2f4 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 @@
 
 }  // 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 80dcf5c..8290265 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 68953a8..2fac8fd 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 @@
 
 }  // 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 97d5d46..491160d 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 8b0fbe1..f526250 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 f5fac73..57b79d6 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 b9c70e9..435c7ae 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 @@
 
 }  // 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 dffed64..038b132 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 5c8ae18..da08f8b 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 @@
 
 }  // 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 9d96d17..7e3bda5 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 d4d0c3b..e26ad61 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 ddafbcd..490c038 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 @@
 
 }  // 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 b84211d..b047fd6 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>