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>
diff --git a/webrtc/modules/audio_coding/main/source/acm_amr.cc b/webrtc/modules/audio_coding/main/source/acm_amr.cc
index 8e8d6d5..5590970 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amr.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_amr.cc
@@ -49,6 +49,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_AMR
ACMAMR::ACMAMR(int16_t /* codec_id */)
: encoder_inst_ptr_(NULL),
@@ -421,4 +423,8 @@
}
#endif
-}
+
+} // namespace acm1
+
+} // namespace webrtc
+
diff --git a/webrtc/modules/audio_coding/main/source/acm_amr.h b/webrtc/modules/audio_coding/main/source/acm_amr.h
index 72ed0a2..19c6572 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amr.h
+++ b/webrtc/modules/audio_coding/main/source/acm_amr.h
@@ -19,7 +19,7 @@
namespace webrtc {
-enum ACMAMRPackingFormat;
+namespace acm1 {
class ACMAMR : public ACMGenericCodec {
public:
@@ -80,6 +80,8 @@
ACMAMRPackingFormat decoder_packing_format_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
index fb86a3b..e2b7635 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
@@ -46,6 +46,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_AMRWB
ACMAMRwb::ACMAMRwb(int16_t /* codec_id */)
: encoder_inst_ptr_(NULL),
@@ -429,4 +431,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_amrwb.h b/webrtc/modules/audio_coding/main/source/acm_amrwb.h
index 485f139..2593418 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amrwb.h
+++ b/webrtc/modules/audio_coding/main/source/acm_amrwb.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMAMRwb : public ACMGenericCodec {
public:
explicit ACMAMRwb(int16_t codec_id);
@@ -81,6 +83,8 @@
ACMAMRPackingFormat decoder_packing_format_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_celt.cc b/webrtc/modules/audio_coding/main/source/acm_celt.cc
index 31d9e37..81a0346 100644
--- a/webrtc/modules/audio_coding/main/source/acm_celt.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_celt.cc
@@ -24,6 +24,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_CELT
ACMCELT::ACMCELT(int16_t /* codec_id */)
@@ -332,4 +334,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_celt.h b/webrtc/modules/audio_coding/main/source/acm_celt.h
index 0bc1afe..4a4610e 100644
--- a/webrtc/modules/audio_coding/main/source/acm_celt.h
+++ b/webrtc/modules/audio_coding/main/source/acm_celt.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMCELT : public ACMGenericCodec {
public:
explicit ACMCELT(int16_t codec_id);
@@ -70,6 +72,8 @@
uint16_t dec_channels_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_cng.cc b/webrtc/modules/audio_coding/main/source/acm_cng.cc
index 0c44aa0..57c48cd 100644
--- a/webrtc/modules/audio_coding/main/source/acm_cng.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_cng.cc
@@ -20,6 +20,8 @@
namespace webrtc {
+namespace acm1 {
+
ACMCNG::ACMCNG(int16_t codec_id) {
encoder_inst_ptr_ = NULL;
decoder_inst_ptr_ = NULL;
@@ -143,4 +145,6 @@
int16_t ACMCNG::EnableDTX() { return -1; }
int16_t ACMCNG::DisableDTX() { return -1; }
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_cng.h b/webrtc/modules/audio_coding/main/source/acm_cng.h
index 32df580..728312d 100644
--- a/webrtc/modules/audio_coding/main/source/acm_cng.h
+++ b/webrtc/modules/audio_coding/main/source/acm_cng.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMCNG: public ACMGenericCodec {
public:
explicit ACMCNG(int16_t codec_id);
@@ -64,6 +66,8 @@
uint16_t samp_freq_hz_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
index 591f74c..c3a54d9 100644
--- a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
@@ -101,6 +101,8 @@
namespace webrtc {
+namespace acm1 {
+
// Not yet used payload-types.
// 83, 82, 81, 80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68,
// 67, 66, 65
@@ -949,4 +951,6 @@
return true;
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.h b/webrtc/modules/audio_coding/main/source/acm_codec_database.h
index da42a6f..7a7054d 100644
--- a/webrtc/modules/audio_coding/main/source/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.h
@@ -22,6 +22,8 @@
namespace webrtc {
+namespace acm1 {
+
// TODO(tlegrand): replace class ACMCodecDB with a namespace.
class ACMCodecDB {
public:
@@ -327,6 +329,8 @@
static const WebRtcNetEQDecoder neteq_decoders_[kMaxNumCodecs];
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_common_defs.h b/webrtc/modules/audio_coding/main/source/acm_common_defs.h
index b959eeb..ecc41f8 100644
--- a/webrtc/modules/audio_coding/main/source/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/main/source/acm_common_defs.h
@@ -26,6 +26,8 @@
namespace webrtc {
+namespace acm1 {
+
// 60 ms is the maximum block size we support. An extra 20 ms is considered
// for safety if process() method is not called when it should be, i.e. we
// accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples.
@@ -104,6 +106,8 @@
uint32_t last_in_timestamp;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc
index 5820bc4..edb6298 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc
@@ -14,6 +14,8 @@
namespace webrtc {
+namespace acm1 {
+
ACMDTMFDetection::ACMDTMFDetection() {}
ACMDTMFDetection::~ACMDTMFDetection() {}
@@ -35,4 +37,6 @@
return -1;
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h
index 43a9047..7455310 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h
@@ -17,6 +17,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMDTMFDetection {
public:
ACMDTMFDetection();
@@ -33,6 +35,8 @@
ACMResampler resampler_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
index 6b91db9..c8dea71 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
@@ -18,6 +18,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_AVT
ACMDTMFPlayout::ACMDTMFPlayout(
@@ -164,4 +166,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h
index 11af234..46175f5 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h
@@ -15,6 +15,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMDTMFPlayout: public ACMGenericCodec {
public:
explicit ACMDTMFPlayout(int16_t codec_id);
@@ -53,6 +55,8 @@
virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g722.cc b/webrtc/modules/audio_coding/main/source/acm_g722.cc
index 1a023db..5368b35 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g722.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g722.cc
@@ -20,6 +20,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_G722
ACMG722::ACMG722(int16_t /* codec_id */)
@@ -351,4 +353,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g722.h b/webrtc/modules/audio_coding/main/source/acm_g722.h
index 8dea5a7..cf7ebe1 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g722.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g722.h
@@ -18,6 +18,8 @@
namespace webrtc {
+namespace acm1 {
+
// forward declaration
struct ACMG722EncStr;
struct ACMG722DecStr;
@@ -75,6 +77,8 @@
G722DecInst* decoder_inst_ptr_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221.cc b/webrtc/modules/audio_coding/main/source/acm_g7221.cc
index f784b62..c9074ac 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221.cc
@@ -86,6 +86,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_G722_1
ACMG722_1::ACMG722_1(int16_t /* codec_id */)
@@ -493,4 +495,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221.h b/webrtc/modules/audio_coding/main/source/acm_g7221.h
index 4e35476..8ea6674 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221.h
@@ -24,6 +24,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMG722_1: public ACMGenericCodec {
public:
explicit ACMG722_1(int16_t codec_id);
@@ -77,6 +79,8 @@
G722_1_32_decinst_t_* decoder_inst32_ptr_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
index a0d9483..91071e9 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
@@ -87,6 +87,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_G722_1C
ACMG722_1C::ACMG722_1C(int16_t /* codec_id */)
@@ -503,4 +505,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221c.h b/webrtc/modules/audio_coding/main/source/acm_g7221c.h
index 1b4e756..d8875aa 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221c.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221c.h
@@ -24,6 +24,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMG722_1C : public ACMGenericCodec {
public:
explicit ACMG722_1C(int16_t codec_id);
@@ -85,6 +87,8 @@
G722_1C_48_decinst_t_* decoder_inst48_ptr_;
};
+} // namespace acm1
+
} // namespace webrtc;
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g729.cc b/webrtc/modules/audio_coding/main/source/acm_g729.cc
index 67611cb..5b75ab9 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g729.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g729.cc
@@ -25,6 +25,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_G729
ACMG729::ACMG729(int16_t /* codec_id */)
@@ -359,4 +361,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g729.h b/webrtc/modules/audio_coding/main/source/acm_g729.h
index d50aa5f..5cfff63 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g729.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g729.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMG729 : public ACMGenericCodec {
public:
explicit ACMG729(int16_t codec_id);
@@ -67,6 +69,8 @@
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7291.cc b/webrtc/modules/audio_coding/main/source/acm_g7291.cc
index da473ca..fd241b3 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7291.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7291.cc
@@ -24,6 +24,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_G729_1
ACMG729_1::ACMG729_1(int16_t /* codec_id */)
@@ -342,4 +344,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7291.h b/webrtc/modules/audio_coding/main/source/acm_g7291.h
index 433b2fd..bac7faf 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7291.h
+++ b/webrtc/modules/audio_coding/main/source/acm_g7291.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMG729_1 : public ACMGenericCodec {
public:
explicit ACMG729_1(int16_t codec_id);
@@ -63,6 +65,8 @@
int16_t flag_g729_mode_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
index 94aeb48..52f5114 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
@@ -22,6 +22,8 @@
namespace webrtc {
+namespace acm1 {
+
// Enum for CNG
enum {
kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER,
@@ -1251,4 +1253,6 @@
bool ACMGenericCodec::IsTrueStereoCodec() { return false; }
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
index 9ba8d08..3951a94 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
@@ -27,6 +27,9 @@
// forward declaration
struct CodecInst;
+
+namespace acm1 {
+
class ACMNetEQ;
class ACMGenericCodec {
@@ -1213,6 +1216,8 @@
uint32_t unique_id_;
};
-} // namespace webrt
+} // namespace acm1
+
+} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
index 22bbbd8..9fa0410 100644
--- a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
@@ -24,6 +24,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_GSMFR
ACMGSMFR::ACMGSMFR(int16_t /* codec_id */)
@@ -260,4 +262,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_gsmfr.h b/webrtc/modules/audio_coding/main/source/acm_gsmfr.h
index 61f5768..aa49973 100644
--- a/webrtc/modules/audio_coding/main/source/acm_gsmfr.h
+++ b/webrtc/modules/audio_coding/main/source/acm_gsmfr.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMGSMFR : public ACMGenericCodec {
public:
explicit ACMGSMFR(int16_t codec_id);
@@ -62,6 +64,8 @@
GSMFR_decinst_t_* decoder_inst_ptr_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
index a2a294e..b47e750 100644
--- a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
@@ -21,6 +21,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_ILBC
ACMILBC::ACMILBC(int16_t /* codec_id */)
@@ -252,4 +254,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_ilbc.h b/webrtc/modules/audio_coding/main/source/acm_ilbc.h
index 50b6df9..bd2495f 100644
--- a/webrtc/modules/audio_coding/main/source/acm_ilbc.h
+++ b/webrtc/modules/audio_coding/main/source/acm_ilbc.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMILBC : public ACMGenericCodec {
public:
explicit ACMILBC(int16_t codec_id);
@@ -62,6 +64,8 @@
iLBC_decinst_t_* decoder_inst_ptr_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_isac.cc b/webrtc/modules/audio_coding/main/source/acm_isac.cc
index e22d3f6..b9316d6 100644
--- a/webrtc/modules/audio_coding/main/source/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_isac.cc
@@ -28,6 +28,8 @@
namespace webrtc {
+namespace acm1 {
+
// we need this otherwise we cannot use forward declaration
// in the header file
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
@@ -896,4 +898,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_isac.h b/webrtc/modules/audio_coding/main/source/acm_isac.h
index 9588723..20b6c53 100644
--- a/webrtc/modules/audio_coding/main/source/acm_isac.h
+++ b/webrtc/modules/audio_coding/main/source/acm_isac.h
@@ -15,6 +15,8 @@
namespace webrtc {
+namespace acm1 {
+
struct ACMISACInst;
enum IsacCodingMode {
@@ -129,6 +131,8 @@
WebRtcACMCodecParams decoder_params_32khz_;
};
-} // namespace
+} // namespace acm1
+
+} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_isac_macros.h b/webrtc/modules/audio_coding/main/source/acm_isac_macros.h
index 6ae4526..01e1e44 100644
--- a/webrtc/modules/audio_coding/main/source/acm_isac_macros.h
+++ b/webrtc/modules/audio_coding/main/source/acm_isac_macros.h
@@ -15,6 +15,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifdef WEBRTC_CODEC_ISAC
#define ACM_ISAC_CREATE WebRtcIsac_Create
#define ACM_ISAC_FREE WebRtcIsac_Free
@@ -67,7 +69,9 @@
#define ACM_ISAC_GETDECSAMPRATE ACMISACFixGetDecSampRate // local Impl
#endif
-} // namespace
+} // namespace acm1
+
+} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq.cc b/webrtc/modules/audio_coding/main/source/acm_neteq.cc
index 5418d18..2ade7bf 100644
--- a/webrtc/modules/audio_coding/main/source/acm_neteq.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_neteq.cc
@@ -26,6 +26,8 @@
namespace webrtc {
+namespace acm1 {
+
#define RTP_HEADER_SIZE 12
#define NETEQ_INIT_FREQ 8000
#define NETEQ_INIT_FREQ_KHZ (NETEQ_INIT_FREQ/1000)
@@ -1140,4 +1142,6 @@
return true;
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq.h b/webrtc/modules/audio_coding/main/source/acm_neteq.h
index 511968b..e52ddc7 100644
--- a/webrtc/modules/audio_coding/main/source/acm_neteq.h
+++ b/webrtc/modules/audio_coding/main/source/acm_neteq.h
@@ -12,8 +12,6 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
#include "webrtc/modules/interface/module_common_types.h"
@@ -25,6 +23,8 @@
class RWLockWrapper;
struct CodecInst;
+namespace acm1 {
+
#define MAX_NUM_SLAVE_NETEQ 1
class ACMNetEQ {
@@ -392,6 +392,8 @@
int maximum_delay_ms_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc b/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc
index aef0acd..8b973ba 100644
--- a/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc
@@ -24,6 +24,8 @@
namespace webrtc {
+namespace acm1 {
+
class AcmNetEqTest : public ::testing::Test {
protected:
static const size_t kMaxPayloadLen = 5760; // 60 ms, 48 kHz, 16 bit samples.
@@ -146,4 +148,6 @@
EXPECT_EQ(-1, stats.medianWaitingTimeMs);
}
-} // namespace
+} // namespace acm1
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.cc b/webrtc/modules/audio_coding/main/source/acm_opus.cc
index 8ea5d51..3a619d0 100644
--- a/webrtc/modules/audio_coding/main/source/acm_opus.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_opus.cc
@@ -23,6 +23,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_OPUS
ACMOpus::ACMOpus(int16_t /* codec_id */)
@@ -312,4 +314,6 @@
#endif // WEBRTC_CODEC_OPUS
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.h b/webrtc/modules/audio_coding/main/source/acm_opus.h
index fa188a1..1e586ff 100644
--- a/webrtc/modules/audio_coding/main/source/acm_opus.h
+++ b/webrtc/modules/audio_coding/main/source/acm_opus.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMOpus : public ACMGenericCodec {
public:
explicit ACMOpus(int16_t codec_id);
@@ -69,6 +71,8 @@
int channels_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
index 91cb9e0..b0032b8 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
@@ -23,6 +23,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_PCM16
ACMPCM16B::ACMPCM16B(int16_t /* codec_id */) {
@@ -244,4 +246,6 @@
}
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcm16b.h b/webrtc/modules/audio_coding/main/source/acm_pcm16b.h
index 38de343..a97589b 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcm16b.h
+++ b/webrtc/modules/audio_coding/main/source/acm_pcm16b.h
@@ -15,6 +15,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMPCM16B : public ACMGenericCodec {
public:
explicit ACMPCM16B(int16_t codec_id);
@@ -58,6 +60,8 @@
int32_t sampling_freq_hz_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcma.cc b/webrtc/modules/audio_coding/main/source/acm_pcma.cc
index 83c1249..c646417 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcma.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcma.cc
@@ -21,6 +21,8 @@
namespace webrtc {
+namespace acm1 {
+
ACMPCMA::ACMPCMA(int16_t codec_id) {
codec_id_ = codec_id;
}
@@ -127,4 +129,6 @@
}
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcma.h b/webrtc/modules/audio_coding/main/source/acm_pcma.h
index 2fc4ea4..cb506ea 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcma.h
+++ b/webrtc/modules/audio_coding/main/source/acm_pcma.h
@@ -15,6 +15,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMPCMA : public ACMGenericCodec {
public:
explicit ACMPCMA(int16_t codec_id);
@@ -56,6 +58,8 @@
int32_t* payload_length) OVERRIDE;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
index 61a64ac..5b6a457 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
@@ -21,6 +21,8 @@
namespace webrtc {
+namespace acm1 {
+
ACMPCMU::ACMPCMU(int16_t codec_id) {
codec_id_ = codec_id;
}
@@ -129,4 +131,6 @@
}
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcmu.h b/webrtc/modules/audio_coding/main/source/acm_pcmu.h
index 309d318..ea401d5 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcmu.h
+++ b/webrtc/modules/audio_coding/main/source/acm_pcmu.h
@@ -15,6 +15,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMPCMU : public ACMGenericCodec {
public:
explicit ACMPCMU(int16_t codec_id);
@@ -56,6 +58,8 @@
int32_t* payload_length) OVERRIDE;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_red.cc b/webrtc/modules/audio_coding/main/source/acm_red.cc
index 6e7ae9f..bc44c72 100644
--- a/webrtc/modules/audio_coding/main/source/acm_red.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_red.cc
@@ -18,6 +18,8 @@
namespace webrtc {
+namespace acm1 {
+
ACMRED::ACMRED(int16_t codec_id) {
codec_id_ = codec_id;
}
@@ -101,4 +103,6 @@
return;
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_red.h b/webrtc/modules/audio_coding/main/source/acm_red.h
index c7bad41..ede18b5 100644
--- a/webrtc/modules/audio_coding/main/source/acm_red.h
+++ b/webrtc/modules/audio_coding/main/source/acm_red.h
@@ -15,6 +15,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMRED : public ACMGenericCodec {
public:
explicit ACMRED(int16_t codec_id);
@@ -53,6 +55,8 @@
virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_resampler.cc b/webrtc/modules/audio_coding/main/source/acm_resampler.cc
index 034dbe5..50ddab1 100644
--- a/webrtc/modules/audio_coding/main/source/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_resampler.cc
@@ -17,6 +17,8 @@
namespace webrtc {
+namespace acm1 {
+
ACMResampler::ACMResampler() {
}
@@ -56,4 +58,6 @@
return out_length / num_audio_channels;
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_resampler.h b/webrtc/modules/audio_coding/main/source/acm_resampler.h
index c23abb8..b50e722 100644
--- a/webrtc/modules/audio_coding/main/source/acm_resampler.h
+++ b/webrtc/modules/audio_coding/main/source/acm_resampler.h
@@ -16,6 +16,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMResampler {
public:
ACMResampler();
@@ -31,6 +33,8 @@
PushResampler resampler_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_speex.cc b/webrtc/modules/audio_coding/main/source/acm_speex.cc
index ce20526..5752693 100644
--- a/webrtc/modules/audio_coding/main/source/acm_speex.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_speex.cc
@@ -25,6 +25,8 @@
namespace webrtc {
+namespace acm1 {
+
#ifndef WEBRTC_CODEC_SPEEX
ACMSPEEX::ACMSPEEX(int16_t /* codec_id */)
: encoder_inst_ptr_(NULL),
@@ -464,4 +466,6 @@
#endif
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_speex.h b/webrtc/modules/audio_coding/main/source/acm_speex.h
index 0f62ea3..762aea8 100644
--- a/webrtc/modules/audio_coding/main/source/acm_speex.h
+++ b/webrtc/modules/audio_coding/main/source/acm_speex.h
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
class ACMSPEEX : public ACMGenericCodec {
public:
explicit ACMSPEEX(int16_t codec_id);
@@ -77,6 +79,8 @@
uint16_t samples_in_20ms_audio_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
index 4ac40b7..9461a1f 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
@@ -21,31 +21,31 @@
// Create module
AudioCodingModule* AudioCodingModule::Create(const int32_t id) {
- return new AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
+ return new acm1::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
}
// Used for testing by inserting a simulated clock. ACM will not destroy the
// injected |clock| the client has to take care of that.
AudioCodingModule* AudioCodingModule::Create(const int32_t id,
Clock* clock) {
- return new AudioCodingModuleImpl(id, clock);
+ return new acm1::AudioCodingModuleImpl(id, clock);
}
// Destroy module
void AudioCodingModule::Destroy(AudioCodingModule* module) {
- delete static_cast<AudioCodingModuleImpl*>(module);
+ delete static_cast<acm1::AudioCodingModuleImpl*>(module);
}
// Get number of supported codecs
uint8_t AudioCodingModule::NumberOfCodecs() {
- return static_cast<uint8_t>(ACMCodecDB::kNumCodecs);
+ return static_cast<uint8_t>(acm1::ACMCodecDB::kNumCodecs);
}
// Get supported codec param with id
int32_t AudioCodingModule::Codec(uint8_t list_id,
CodecInst* codec) {
// Get the codec settings for the codec with the given list ID
- return ACMCodecDB::Codec(list_id, codec);
+ return acm1::ACMCodecDB::Codec(list_id, codec);
}
// Get supported codec Param with name, frequency and number of channels.
@@ -55,7 +55,8 @@
int codec_id;
// Get the id of the codec from the database.
- codec_id = ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
+ codec_id = acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz,
+ channels);
if (codec_id < 0) {
// We couldn't find a matching codec, set the parameters to unacceptable
// values and return.
@@ -68,7 +69,7 @@
}
// Get default codec settings.
- ACMCodecDB::Codec(codec_id, codec);
+ acm1::ACMCodecDB::Codec(codec_id, codec);
// Keep the number of channels from the function call. For most codecs it
// will be the same value as in default codec settings, but not for all.
@@ -80,14 +81,14 @@
// Get supported codec Index with name, frequency and number of channels.
int32_t AudioCodingModule::Codec(const char* payload_name,
int sampling_freq_hz, int channels) {
- return ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
+ return acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
}
// Checks the validity of the parameters of the given codec
bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
int mirror_id;
- int codec_number = ACMCodecDB::CodecNumber(&codec, &mirror_id);
+ int codec_number = acm1::ACMCodecDB::CodecNumber(&codec, &mirror_id);
if (codec_number < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
@@ -99,8 +100,8 @@
}
AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
- return new AudioCodingModuleImpl(static_cast<int32_t>(id),
- Clock::GetRealTimeClock());
+ return new acm1::AudioCodingModuleImpl(static_cast<int32_t>(id),
+ Clock::GetRealTimeClock());
}
AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index b136d84..93b21e6 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -32,6 +32,8 @@
namespace webrtc {
+namespace acm1 {
+
enum {
kACMToneEnd = 999
};
@@ -3115,4 +3117,6 @@
nack_enabled_ = false;
}
+} // namespace acm1
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
index f58e3e5..64afe4f 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -15,6 +15,7 @@
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
@@ -22,11 +23,16 @@
namespace webrtc {
-class ACMDTMFDetection;
-class ACMGenericCodec;
class CriticalSectionWrapper;
class RWLockWrapper;
class Clock;
+
+namespace acm1 {
+
+struct WebRtcACMAudioBuff;
+struct WebRtcACMCodecParams;
+class ACMDTMFDetection;
+class ACMGenericCodec;
class Nack;
class AudioCodingModuleImpl : public AudioCodingModule {
@@ -88,8 +94,7 @@
// Register a transport callback which will be
// called to deliver the encoded buffers.
- int32_t RegisterTransportCallback(
- AudioPacketizationCallback* transport);
+ int32_t RegisterTransportCallback(AudioPacketizationCallback* transport);
// Used by the module to deliver messages to the codec module/application
// AVT(DTMF).
@@ -125,8 +130,7 @@
bool enable_vad = false,
ACMVADMode mode = VADNormal);
- int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
- ACMVADMode* mode) const;
+ int32_t VAD(bool* dtx_enabled, bool* vad_enabled, ACMVADMode* mode) const;
int32_t RegisterVADCallback(ACMVADCallback* vad_callback);
@@ -454,6 +458,8 @@
bool nack_enabled_;
};
+} // namespace acm1
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/webrtc/modules/audio_coding/main/source/nack.cc b/webrtc/modules/audio_coding/main/source/nack.cc
index ec6cb3d..4ca260d 100644
--- a/webrtc/modules/audio_coding/main/source/nack.cc
+++ b/webrtc/modules/audio_coding/main/source/nack.cc
@@ -19,6 +19,8 @@
namespace webrtc {
+namespace acm1 {
+
namespace {
const int kDefaultSampleRateKhz = 48;
@@ -222,4 +224,6 @@
return sequence_numbers;
}
-} // webrtc
+} // namespace acm1
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/nack.h b/webrtc/modules/audio_coding/main/source/nack.h
index e047c28..9cea15d 100644
--- a/webrtc/modules/audio_coding/main/source/nack.h
+++ b/webrtc/modules/audio_coding/main/source/nack.h
@@ -49,6 +49,8 @@
//
namespace webrtc {
+namespace acm1 {
+
class Nack {
public:
// A limit for the size of the NACK list.
@@ -204,6 +206,8 @@
size_t max_nack_list_size_;
};
-} // webrtc
+} // namespace acm1
+
+} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_NACK_H_
diff --git a/webrtc/modules/audio_coding/main/source/nack_unittest.cc b/webrtc/modules/audio_coding/main/source/nack_unittest.cc
index ba92f0e..811aca4 100644
--- a/webrtc/modules/audio_coding/main/source/nack_unittest.cc
+++ b/webrtc/modules/audio_coding/main/source/nack_unittest.cc
@@ -13,6 +13,7 @@
#include <stdint.h>
#include <algorithm>
+#include <vector>
#include "gtest/gtest.h"
#include "webrtc/typedefs.h"
@@ -21,6 +22,8 @@
namespace webrtc {
+namespace acm1 {
+
namespace {
const int kNackThreshold = 3;
@@ -479,4 +482,6 @@
EXPECT_EQ(5, nack_list[1]);
}
-} // webrtc
+} // namespace acm1
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/main/test/opus_test.h
index ca0da9e..49b98ea 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.h
+++ b/webrtc/modules/audio_coding/main/test/opus_test.h
@@ -29,6 +29,7 @@
~OpusTest();
void Perform();
+
private:
void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
int percent_loss = 0);
@@ -44,7 +45,7 @@
int counter_;
uint8_t payload_type_;
int rtp_timestamp_;
- ACMResampler resampler_;
+ acm1::ACMResampler resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
WebRtcOpusDecInst* opus_mono_decoder_;