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

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"
 

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_

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"
 

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_

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
 

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_

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 {

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_

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 {

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_

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_

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
 

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_

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
 

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_

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.

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_

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.

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_

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
 

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_

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
 

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_

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 {

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_

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
 

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_

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
 

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_

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 {

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_

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_
 

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
 

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_

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
 

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_

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

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_

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.

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_

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"

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_

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"

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 {

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_

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>
 

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_

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
 

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_

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 {

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
-     ],
-    }],
-  ],
 }

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"

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_

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 {
 

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_

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 {
 

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.
 

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_

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>
 

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
+

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_

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

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_

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

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_

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

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_

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

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_

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_

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

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_

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

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_

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

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_

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

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_

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

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_

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

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_

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

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_

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

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_

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

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_

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

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_

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

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_

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_
 

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

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_

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

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

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_

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

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_