From 48af652ea5687bd874049c9ac695ae3872adfd58 Mon Sep 17 00:00:00 2001 From: "turaj@webrtc.org" Date: Fri, 13 Sep 2013 23:06:59 +0000 Subject: [PATCH] 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 --- .../modules/audio_coding/main/acm2/acm_amr.cc | 4 +- .../modules/audio_coding/main/acm2/acm_amr.h | 8 +- .../audio_coding/main/acm2/acm_amrwb.cc | 4 +- .../audio_coding/main/acm2/acm_amrwb.h | 8 +- .../audio_coding/main/acm2/acm_celt.cc | 4 +- .../modules/audio_coding/main/acm2/acm_celt.h | 8 +- .../modules/audio_coding/main/acm2/acm_cng.cc | 6 +- .../modules/audio_coding/main/acm2/acm_cng.h | 8 +- .../main/acm2/acm_codec_database.cc | 44 ++++---- .../main/acm2/acm_codec_database.h | 8 +- .../audio_coding/main/acm2/acm_common_defs.h | 6 +- .../main/acm2/acm_dtmf_playout.cc | 6 +- .../audio_coding/main/acm2/acm_dtmf_playout.h | 8 +- .../audio_coding/main/acm2/acm_g722.cc | 6 +- .../modules/audio_coding/main/acm2/acm_g722.h | 8 +- .../audio_coding/main/acm2/acm_g7221.cc | 6 +- .../audio_coding/main/acm2/acm_g7221.h | 8 +- .../audio_coding/main/acm2/acm_g7221c.cc | 6 +- .../audio_coding/main/acm2/acm_g7221c.h | 8 +- .../audio_coding/main/acm2/acm_g729.cc | 6 +- .../modules/audio_coding/main/acm2/acm_g729.h | 8 +- .../audio_coding/main/acm2/acm_g7291.cc | 4 +- .../audio_coding/main/acm2/acm_g7291.h | 8 +- .../main/acm2/acm_generic_codec.cc | 6 +- .../main/acm2/acm_generic_codec.h | 8 +- .../audio_coding/main/acm2/acm_gsmfr.cc | 4 +- .../audio_coding/main/acm2/acm_gsmfr.h | 8 +- .../audio_coding/main/acm2/acm_ilbc.cc | 4 +- .../modules/audio_coding/main/acm2/acm_ilbc.h | 8 +- .../audio_coding/main/acm2/acm_isac.cc | 8 +- .../modules/audio_coding/main/acm2/acm_isac.h | 8 +- .../audio_coding/main/acm2/acm_isac_macros.h | 6 +- .../audio_coding/main/acm2/acm_opus.cc | 6 +- .../modules/audio_coding/main/acm2/acm_opus.h | 8 +- .../audio_coding/main/acm2/acm_pcm16b.cc | 6 +- .../audio_coding/main/acm2/acm_pcm16b.h | 8 +- .../audio_coding/main/acm2/acm_pcma.cc | 4 +- .../modules/audio_coding/main/acm2/acm_pcma.h | 8 +- .../audio_coding/main/acm2/acm_pcmu.cc | 4 +- .../modules/audio_coding/main/acm2/acm_pcmu.h | 8 +- .../audio_coding/main/acm2/acm_receiver.cc | 8 +- .../audio_coding/main/acm2/acm_receiver.h | 12 +-- .../main/acm2/acm_receiver_unittest.cc | 4 +- .../modules/audio_coding/main/acm2/acm_red.cc | 4 +- .../modules/audio_coding/main/acm2/acm_red.h | 8 +- .../audio_coding/main/acm2/acm_resampler.cc | 2 +- .../audio_coding/main/acm2/acm_resampler.h | 6 +- .../audio_coding/main/acm2/acm_speex.cc | 6 +- .../audio_coding/main/acm2/acm_speex.h | 8 +- .../main/acm2/audio_coding_module.cc | 4 +- .../main/acm2/audio_coding_module.gypi | 102 +----------------- .../main/acm2/audio_coding_module_impl.cc | 10 +- .../main/acm2/audio_coding_module_impl.h | 12 +-- .../main/acm2/initial_delay_manager.cc | 2 +- .../main/acm2/initial_delay_manager.h | 6 +- .../acm2/initial_delay_manager_unittest.cc | 2 +- webrtc/modules/audio_coding/main/acm2/nack.cc | 2 +- webrtc/modules/audio_coding/main/acm2/nack.h | 6 +- .../audio_coding/main/acm2/nack_unittest.cc | 2 +- .../audio_coding/main/source/acm_amr.cc | 8 +- .../audio_coding/main/source/acm_amr.h | 4 +- .../audio_coding/main/source/acm_amrwb.cc | 4 + .../audio_coding/main/source/acm_amrwb.h | 4 + .../audio_coding/main/source/acm_celt.cc | 4 + .../audio_coding/main/source/acm_celt.h | 4 + .../audio_coding/main/source/acm_cng.cc | 4 + .../audio_coding/main/source/acm_cng.h | 4 + .../main/source/acm_codec_database.cc | 4 + .../main/source/acm_codec_database.h | 4 + .../main/source/acm_common_defs.h | 4 + .../main/source/acm_dtmf_detection.cc | 4 + .../main/source/acm_dtmf_detection.h | 4 + .../main/source/acm_dtmf_playout.cc | 4 + .../main/source/acm_dtmf_playout.h | 4 + .../audio_coding/main/source/acm_g722.cc | 4 + .../audio_coding/main/source/acm_g722.h | 4 + .../audio_coding/main/source/acm_g7221.cc | 4 + .../audio_coding/main/source/acm_g7221.h | 4 + .../audio_coding/main/source/acm_g7221c.cc | 4 + .../audio_coding/main/source/acm_g7221c.h | 4 + .../audio_coding/main/source/acm_g729.cc | 4 + .../audio_coding/main/source/acm_g729.h | 4 + .../audio_coding/main/source/acm_g7291.cc | 4 + .../audio_coding/main/source/acm_g7291.h | 4 + .../main/source/acm_generic_codec.cc | 4 + .../main/source/acm_generic_codec.h | 7 +- .../audio_coding/main/source/acm_gsmfr.cc | 4 + .../audio_coding/main/source/acm_gsmfr.h | 4 + .../audio_coding/main/source/acm_ilbc.cc | 4 + .../audio_coding/main/source/acm_ilbc.h | 4 + .../audio_coding/main/source/acm_isac.cc | 4 + .../audio_coding/main/source/acm_isac.h | 6 +- .../main/source/acm_isac_macros.h | 6 +- .../audio_coding/main/source/acm_neteq.cc | 4 + .../audio_coding/main/source/acm_neteq.h | 6 +- .../main/source/acm_neteq_unittest.cc | 6 +- .../audio_coding/main/source/acm_opus.cc | 4 + .../audio_coding/main/source/acm_opus.h | 4 + .../audio_coding/main/source/acm_pcm16b.cc | 4 + .../audio_coding/main/source/acm_pcm16b.h | 4 + .../audio_coding/main/source/acm_pcma.cc | 4 + .../audio_coding/main/source/acm_pcma.h | 4 + .../audio_coding/main/source/acm_pcmu.cc | 4 + .../audio_coding/main/source/acm_pcmu.h | 4 + .../audio_coding/main/source/acm_red.cc | 4 + .../audio_coding/main/source/acm_red.h | 4 + .../audio_coding/main/source/acm_resampler.cc | 4 + .../audio_coding/main/source/acm_resampler.h | 4 + .../audio_coding/main/source/acm_speex.cc | 4 + .../audio_coding/main/source/acm_speex.h | 4 + .../main/source/audio_coding_module.cc | 23 ++-- .../main/source/audio_coding_module_impl.cc | 4 + .../main/source/audio_coding_module_impl.h | 18 ++-- .../modules/audio_coding/main/source/nack.cc | 6 +- .../modules/audio_coding/main/source/nack.h | 6 +- .../audio_coding/main/source/nack_unittest.cc | 7 +- .../audio_coding/main/test/opus_test.h | 3 +- 117 files changed, 461 insertions(+), 333 deletions(-) diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amr.cc b/webrtc/modules/audio_coding/main/acm2/acm_amr.cc index 75430f1a86..ab4003abbf 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 c58b5111f6..4471e6bca7 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_amr.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_amr.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMR_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct AMR_encinst_t_; @@ -62,4 +62,4 @@ class ACMAMR : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMR_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc index 1b82674b1d..849353a933 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 550bab2d32..e5bd99d9bb 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMRWB_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMRWB_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct AMRWB_encinst_t_; @@ -63,4 +63,4 @@ class ACMAMRwb : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMRWB_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_celt.cc b/webrtc/modules/audio_coding/main/acm2/acm_celt.cc index 6f2c807e1b..21fa3a9d0d 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 b90a4e8500..4b40f799e9 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_celt.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_celt.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CELT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CELT_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct CELT_encinst_t_; @@ -47,4 +47,4 @@ class ACMCELT : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CELT_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_cng.cc b/webrtc/modules/audio_coding/main/acm2/acm_cng.cc index b04fd6ad11..9e658bdad1 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 2ea4f02db7..3816fa2a89 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_cng.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_cng.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CNG_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CNG_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct WebRtcCngEncInst; @@ -53,4 +53,4 @@ class ACMCNG: public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CNG_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc index 08080d1dcb..8e14fbbaf6 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 -#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 fb5cb9a039..a8a76438c0 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h @@ -13,11 +13,11 @@ * codecs. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_ #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" namespace webrtc { @@ -347,4 +347,4 @@ class ACMCodecDB { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h index b27256a00f..39287ea626 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 @@ -95,4 +95,4 @@ struct WebRtcACMCodecParams { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc index b6b91029de..ca7e86fd84 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 e16653cdf6..4c3154ca9c 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DTMF_PLAYOUT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DTMF_PLAYOUT_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" namespace webrtc { @@ -37,4 +37,4 @@ class ACMDTMFPlayout : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DTMF_PLAYOUT_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g722.cc b/webrtc/modules/audio_coding/main/acm2/acm_g722.cc index 6ba0d7b4a7..fe2bd6cb9f 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 21a0fdb2e9..34b6c8516c 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_g722.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_g722.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" typedef struct WebRtcG722EncInst G722EncInst; typedef struct WebRtcG722DecInst G722DecInst; @@ -54,4 +54,4 @@ class ACMG722 : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc index 65b34b08fb..0cba710848 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 2b532db93a..4a0bd480d9 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_g7221.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct G722_1_16_encinst_t_; @@ -59,4 +59,4 @@ class ACMG722_1 : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc index b426d1f8b1..531008af27 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 d051b28b6c..961ed4e17a 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221C_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221C_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct G722_1C_24_encinst_t_; @@ -59,4 +59,4 @@ class ACMG722_1C : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221C_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g729.cc b/webrtc/modules/audio_coding/main/acm2/acm_g729.cc index a2349ce4a8..91dbb43ee1 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 3b35f3b90e..f7e762cbaf 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_g729.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_g729.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G729_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G729_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct G729_encinst_t_; @@ -51,4 +51,4 @@ class ACMG729 : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G729_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc index 1c661c11b1..f16eec89b6 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 97601eac1a..5a38e59a34 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_g7291.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_g7291.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7291_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7291_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct G729_1_inst_t_; @@ -49,4 +49,4 @@ class ACMG729_1 : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7291_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc index 5e210efd6d..4c89b108d7 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 #include #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 cde2e42843..0129bf38ba 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_ #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" #include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" @@ -915,4 +915,4 @@ class ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc index 9fd097c2ef..44e6e3d917 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 935ac444fa..51c29eea41 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct GSMFR_encinst_t_; @@ -47,4 +47,4 @@ class ACMGSMFR : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc index 204e1e950f..14fbbd4506 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 11e759c6a1..e02c789d3f 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ILBC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ILBC_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct iLBC_encinst_t_; @@ -45,4 +45,4 @@ class ACMILBC : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ILBC_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc index 1c7e8b3937..e2de7efb27 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 #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 f4cf1a6ac9..2e6657fb40 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_isac.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" namespace webrtc { @@ -95,4 +95,4 @@ class ACMISAC : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h b/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h index 646b3cc380..c2a782095a 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_MACROS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_MACROS_H_ #include "webrtc/engine_configurations.h" @@ -72,5 +72,5 @@ namespace webrtc { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_MACROS_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_opus.cc b/webrtc/modules/audio_coding/main/acm2/acm_opus.cc index a7380859c6..d627fad8d0 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 28b08b6fd3..caac01093a 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_opus.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_opus.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_ #include "webrtc/common_audio/resampler/include/resampler.h" -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" struct WebRtcOpusEncInst; struct WebRtcOpusDecInst; @@ -47,4 +47,4 @@ class ACMOpus : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc index 3bc964241f..7c5b0bd329 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 a7fff0f52b..32490209a2 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCM16B_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCM16B_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" namespace webrtc { @@ -39,4 +39,4 @@ class ACMPCM16B : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCM16B_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc index 0d574fe24b..cb5ebccfdc 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 61386d3127..4102e17d97 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_pcma.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_pcma.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMA_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMA_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" namespace webrtc { @@ -37,4 +37,4 @@ class ACMPCMA : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMA_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc index 441e3ddcde..6f479ed219 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 832a00d1d7..2898df6370 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMU_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMU_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" namespace webrtc { @@ -37,4 +37,4 @@ class ACMPCMU : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMU_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc index fb3fe3e60a..5a36f860bd 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 // 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 344e1c9206..5f6d684b0a 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 #include "webrtc/common_audio/vad/include/webrtc_vad.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/source/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/source/initial_delay_manager.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" @@ -362,4 +362,4 @@ class AcmReceiver { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RECEIVER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc index ab652cfaa4..6fa6743ba0 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 // 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 5b5b16f0de..f4a1f6f2a5 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 c8023db635..ab8d913fa8 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_red.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_red.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RED_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RED_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" namespace webrtc { @@ -37,4 +37,4 @@ class ACMRED : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RED_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc index d399cee8e7..13eed0ba6d 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 diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h index c44fbc47f7..8abb2f4f7c 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/typedefs.h" @@ -37,4 +37,4 @@ class ACMResampler { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_speex.cc b/webrtc/modules/audio_coding/main/acm2/acm_speex.cc index 80dcf5c20a..829026549d 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 68953a8d8d..2fac8fd2e9 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_speex.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_speex.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SPEEX_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SPEEX_H_ -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" // forward declaration struct SPEEX_encinst_t_; @@ -62,4 +62,4 @@ class ACMSPEEX : public ACMGenericCodec { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SPEEX_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc index 97d5d46287..491160d8bd 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 8b0fbe111a..f52625037b 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 f5fac732eb..57b79d61d1 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 #include @@ -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 b9c70e9bb9..435c7aeab8 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 #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/source/acm_receiver.h" -#include "webrtc/modules/audio_coding/main/source/acm_resampler.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { @@ -351,4 +351,4 @@ class AudioCodingModuleImpl : public AudioCodingModule { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc index dffed646c7..038b13272d 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 5c8ae18be5..da08f8bd87 100644 --- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h +++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_INITIAL_DELAY_MANAGER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_INITIAL_DELAY_MANAGER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_ #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" @@ -112,4 +112,4 @@ class InitialDelayManager { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_INITIAL_DELAY_MANAGER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc index 9d96d1770d..7e3bda5b50 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 #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 d4d0c3b8fb..e26ad611f7 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 // For assert. diff --git a/webrtc/modules/audio_coding/main/acm2/nack.h b/webrtc/modules/audio_coding/main/acm2/nack.h index ddafbcdf63..490c038187 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 #include @@ -206,4 +206,4 @@ class Nack { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_NACK_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc b/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc index b84211dc32..b047fd6d01 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 diff --git a/webrtc/modules/audio_coding/main/source/acm_amr.cc b/webrtc/modules/audio_coding/main/source/acm_amr.cc index 8e8d6d51f4..5590970d62 100644 --- a/webrtc/modules/audio_coding/main/source/acm_amr.cc +++ b/webrtc/modules/audio_coding/main/source/acm_amr.cc @@ -49,6 +49,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_AMR ACMAMR::ACMAMR(int16_t /* codec_id */) : encoder_inst_ptr_(NULL), @@ -421,4 +423,8 @@ ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const { } #endif -} + +} // namespace acm1 + +} // namespace webrtc + diff --git a/webrtc/modules/audio_coding/main/source/acm_amr.h b/webrtc/modules/audio_coding/main/source/acm_amr.h index 72ed0a22bd..19c657246a 100644 --- a/webrtc/modules/audio_coding/main/source/acm_amr.h +++ b/webrtc/modules/audio_coding/main/source/acm_amr.h @@ -19,7 +19,7 @@ struct AMR_decinst_t_; namespace webrtc { -enum ACMAMRPackingFormat; +namespace acm1 { class ACMAMR : public ACMGenericCodec { public: @@ -80,6 +80,8 @@ class ACMAMR : public ACMGenericCodec { ACMAMRPackingFormat decoder_packing_format_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc index fb86a3b42f..e2b7635e21 100644 --- a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc +++ b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc @@ -46,6 +46,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_AMRWB ACMAMRwb::ACMAMRwb(int16_t /* codec_id */) : encoder_inst_ptr_(NULL), @@ -429,4 +431,6 @@ ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_amrwb.h b/webrtc/modules/audio_coding/main/source/acm_amrwb.h index 485f1395ab..25934187e5 100644 --- a/webrtc/modules/audio_coding/main/source/acm_amrwb.h +++ b/webrtc/modules/audio_coding/main/source/acm_amrwb.h @@ -19,6 +19,8 @@ struct AMRWB_decinst_t_; namespace webrtc { +namespace acm1 { + class ACMAMRwb : public ACMGenericCodec { public: explicit ACMAMRwb(int16_t codec_id); @@ -81,6 +83,8 @@ class ACMAMRwb : public ACMGenericCodec { ACMAMRPackingFormat decoder_packing_format_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_celt.cc b/webrtc/modules/audio_coding/main/source/acm_celt.cc index 31d9e378f6..81a034686a 100644 --- a/webrtc/modules/audio_coding/main/source/acm_celt.cc +++ b/webrtc/modules/audio_coding/main/source/acm_celt.cc @@ -24,6 +24,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_CELT ACMCELT::ACMCELT(int16_t /* codec_id */) @@ -332,4 +334,6 @@ void ACMCELT::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_celt.h b/webrtc/modules/audio_coding/main/source/acm_celt.h index 0bc1afe13c..4a4610e0d4 100644 --- a/webrtc/modules/audio_coding/main/source/acm_celt.h +++ b/webrtc/modules/audio_coding/main/source/acm_celt.h @@ -19,6 +19,8 @@ struct CELT_decinst_t_; namespace webrtc { +namespace acm1 { + class ACMCELT : public ACMGenericCodec { public: explicit ACMCELT(int16_t codec_id); @@ -70,6 +72,8 @@ class ACMCELT : public ACMGenericCodec { uint16_t dec_channels_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_cng.cc b/webrtc/modules/audio_coding/main/source/acm_cng.cc index 0c44aa0ab3..57c48cd837 100644 --- a/webrtc/modules/audio_coding/main/source/acm_cng.cc +++ b/webrtc/modules/audio_coding/main/source/acm_cng.cc @@ -20,6 +20,8 @@ namespace webrtc { +namespace acm1 { + ACMCNG::ACMCNG(int16_t codec_id) { encoder_inst_ptr_ = NULL; decoder_inst_ptr_ = NULL; @@ -143,4 +145,6 @@ void ACMCNG::InternalDestructEncoderInst(void* ptr_inst) { int16_t ACMCNG::EnableDTX() { return -1; } int16_t ACMCNG::DisableDTX() { return -1; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_cng.h b/webrtc/modules/audio_coding/main/source/acm_cng.h index 32df580603..728312d55f 100644 --- a/webrtc/modules/audio_coding/main/source/acm_cng.h +++ b/webrtc/modules/audio_coding/main/source/acm_cng.h @@ -19,6 +19,8 @@ struct WebRtcCngDecInst; namespace webrtc { +namespace acm1 { + class ACMCNG: public ACMGenericCodec { public: explicit ACMCNG(int16_t codec_id); @@ -64,6 +66,8 @@ class ACMCNG: public ACMGenericCodec { uint16_t samp_freq_hz_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc index 591f74c728..c3a54d922b 100644 --- a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc +++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc @@ -101,6 +101,8 @@ namespace webrtc { +namespace acm1 { + // Not yet used payload-types. // 83, 82, 81, 80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68, // 67, 66, 65 @@ -949,4 +951,6 @@ bool ACMCodecDB::ValidPayloadType(int payload_type) { return true; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.h b/webrtc/modules/audio_coding/main/source/acm_codec_database.h index da42a6fa4c..7a7054dd1a 100644 --- a/webrtc/modules/audio_coding/main/source/acm_codec_database.h +++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.h @@ -22,6 +22,8 @@ namespace webrtc { +namespace acm1 { + // TODO(tlegrand): replace class ACMCodecDB with a namespace. class ACMCodecDB { public: @@ -327,6 +329,8 @@ class ACMCodecDB { static const WebRtcNetEQDecoder neteq_decoders_[kMaxNumCodecs]; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_common_defs.h b/webrtc/modules/audio_coding/main/source/acm_common_defs.h index b959eeb47d..ecc41f8c93 100644 --- a/webrtc/modules/audio_coding/main/source/acm_common_defs.h +++ b/webrtc/modules/audio_coding/main/source/acm_common_defs.h @@ -26,6 +26,8 @@ namespace webrtc { +namespace acm1 { + // 60 ms is the maximum block size we support. An extra 20 ms is considered // for safety if process() method is not called when it should be, i.e. we // accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples. @@ -104,6 +106,8 @@ struct WebRtcACMAudioBuff { uint32_t last_in_timestamp; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc index 5820bc4ab8..edb6298768 100644 --- a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc +++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc @@ -14,6 +14,8 @@ namespace webrtc { +namespace acm1 { + ACMDTMFDetection::ACMDTMFDetection() {} ACMDTMFDetection::~ACMDTMFDetection() {} @@ -35,4 +37,6 @@ int16_t ACMDTMFDetection::Detect( return -1; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h index 43a9047a77..74553107a3 100644 --- a/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h +++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h @@ -17,6 +17,8 @@ namespace webrtc { +namespace acm1 { + class ACMDTMFDetection { public: ACMDTMFDetection(); @@ -33,6 +35,8 @@ class ACMDTMFDetection { ACMResampler resampler_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc index 6b91db98e1..c8dea71825 100644 --- a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc +++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc @@ -18,6 +18,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_AVT ACMDTMFPlayout::ACMDTMFPlayout( @@ -164,4 +166,6 @@ void ACMDTMFPlayout::DestructDecoderSafe() { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h index 11af23495a..46175f59e6 100644 --- a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h +++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h @@ -15,6 +15,8 @@ namespace webrtc { +namespace acm1 { + class ACMDTMFPlayout: public ACMGenericCodec { public: explicit ACMDTMFPlayout(int16_t codec_id); @@ -53,6 +55,8 @@ class ACMDTMFPlayout: public ACMGenericCodec { virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_g722.cc b/webrtc/modules/audio_coding/main/source/acm_g722.cc index 1a023db543..5368b35f94 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g722.cc +++ b/webrtc/modules/audio_coding/main/source/acm_g722.cc @@ -20,6 +20,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_G722 ACMG722::ACMG722(int16_t /* codec_id */) @@ -351,4 +353,6 @@ void ACMG722::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_g722.h b/webrtc/modules/audio_coding/main/source/acm_g722.h index 8dea5a7a5e..cf7ebe1e22 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g722.h +++ b/webrtc/modules/audio_coding/main/source/acm_g722.h @@ -18,6 +18,8 @@ typedef struct WebRtcG722DecInst G722DecInst; namespace webrtc { +namespace acm1 { + // forward declaration struct ACMG722EncStr; struct ACMG722DecStr; @@ -75,6 +77,8 @@ class ACMG722 : public ACMGenericCodec { G722DecInst* decoder_inst_ptr_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221.cc b/webrtc/modules/audio_coding/main/source/acm_g7221.cc index f784b6226e..c9074ac7ce 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g7221.cc +++ b/webrtc/modules/audio_coding/main/source/acm_g7221.cc @@ -86,6 +86,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_G722_1 ACMG722_1::ACMG722_1(int16_t /* codec_id */) @@ -493,4 +495,6 @@ void ACMG722_1::InternalDestructEncoderInst(void* ptr_inst) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221.h b/webrtc/modules/audio_coding/main/source/acm_g7221.h index 4e35476d46..8ea66742c9 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g7221.h +++ b/webrtc/modules/audio_coding/main/source/acm_g7221.h @@ -24,6 +24,8 @@ struct G722_1_Inst_t_; namespace webrtc { +namespace acm1 { + class ACMG722_1: public ACMGenericCodec { public: explicit ACMG722_1(int16_t codec_id); @@ -77,6 +79,8 @@ class ACMG722_1: public ACMGenericCodec { G722_1_32_decinst_t_* decoder_inst32_ptr_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc index a0d94836a8..91071e9b4e 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc +++ b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc @@ -87,6 +87,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_G722_1C ACMG722_1C::ACMG722_1C(int16_t /* codec_id */) @@ -503,4 +505,6 @@ void ACMG722_1C::InternalDestructEncoderInst(void* ptr_inst) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221c.h b/webrtc/modules/audio_coding/main/source/acm_g7221c.h index 1b4e756492..d8875aa2fb 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g7221c.h +++ b/webrtc/modules/audio_coding/main/source/acm_g7221c.h @@ -24,6 +24,8 @@ struct G722_1_Inst_t_; namespace webrtc { +namespace acm1 { + class ACMG722_1C : public ACMGenericCodec { public: explicit ACMG722_1C(int16_t codec_id); @@ -85,6 +87,8 @@ class ACMG722_1C : public ACMGenericCodec { G722_1C_48_decinst_t_* decoder_inst48_ptr_; }; +} // namespace acm1 + } // namespace webrtc; #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_g729.cc b/webrtc/modules/audio_coding/main/source/acm_g729.cc index 67611cbdc1..5b75ab948c 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g729.cc +++ b/webrtc/modules/audio_coding/main/source/acm_g729.cc @@ -25,6 +25,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_G729 ACMG729::ACMG729(int16_t /* codec_id */) @@ -359,4 +361,6 @@ void ACMG729::InternalDestructEncoderInst(void* ptr_inst) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_g729.h b/webrtc/modules/audio_coding/main/source/acm_g729.h index d50aa5f3fa..5cfff63b69 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g729.h +++ b/webrtc/modules/audio_coding/main/source/acm_g729.h @@ -19,6 +19,8 @@ struct G729_decinst_t_; namespace webrtc { +namespace acm1 { + class ACMG729 : public ACMGenericCodec { public: explicit ACMG729(int16_t codec_id); @@ -67,6 +69,8 @@ class ACMG729 : public ACMGenericCodec { }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_g7291.cc b/webrtc/modules/audio_coding/main/source/acm_g7291.cc index da473ca84b..fd241b3935 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g7291.cc +++ b/webrtc/modules/audio_coding/main/source/acm_g7291.cc @@ -24,6 +24,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_G729_1 ACMG729_1::ACMG729_1(int16_t /* codec_id */) @@ -342,4 +344,6 @@ int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_g7291.h b/webrtc/modules/audio_coding/main/source/acm_g7291.h index 433b2fdf89..bac7faf836 100644 --- a/webrtc/modules/audio_coding/main/source/acm_g7291.h +++ b/webrtc/modules/audio_coding/main/source/acm_g7291.h @@ -19,6 +19,8 @@ struct G729_1_inst_t_; namespace webrtc { +namespace acm1 { + class ACMG729_1 : public ACMGenericCodec { public: explicit ACMG729_1(int16_t codec_id); @@ -63,6 +65,8 @@ class ACMG729_1 : public ACMGenericCodec { int16_t flag_g729_mode_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc index 94aeb48372..52f51146b0 100644 --- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc +++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc @@ -22,6 +22,8 @@ namespace webrtc { +namespace acm1 { + // Enum for CNG enum { kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER, @@ -1251,4 +1253,6 @@ int16_t ACMGenericCodec::REDPayloadISAC(const int32_t /* isac_rate */, bool ACMGenericCodec::IsTrueStereoCodec() { return false; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h index 9ba8d08fd8..3951a94d54 100644 --- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h +++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h @@ -27,6 +27,9 @@ namespace webrtc { // forward declaration struct CodecInst; + +namespace acm1 { + class ACMNetEQ; class ACMGenericCodec { @@ -1213,6 +1216,8 @@ class ACMGenericCodec { uint32_t unique_id_; }; -} // namespace webrt +} // namespace acm1 + +} // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc index 22bbbd8f8b..9fa0410645 100644 --- a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc +++ b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc @@ -24,6 +24,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_GSMFR ACMGSMFR::ACMGSMFR(int16_t /* codec_id */) @@ -260,4 +262,6 @@ void ACMGSMFR::InternalDestructEncoderInst(void* ptr_inst) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_gsmfr.h b/webrtc/modules/audio_coding/main/source/acm_gsmfr.h index 61f576891f..aa499734af 100644 --- a/webrtc/modules/audio_coding/main/source/acm_gsmfr.h +++ b/webrtc/modules/audio_coding/main/source/acm_gsmfr.h @@ -19,6 +19,8 @@ struct GSMFR_decinst_t_; namespace webrtc { +namespace acm1 { + class ACMGSMFR : public ACMGenericCodec { public: explicit ACMGSMFR(int16_t codec_id); @@ -62,6 +64,8 @@ class ACMGSMFR : public ACMGenericCodec { GSMFR_decinst_t_* decoder_inst_ptr_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc index a2a294ef5b..b47e75090d 100644 --- a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc +++ b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc @@ -21,6 +21,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_ILBC ACMILBC::ACMILBC(int16_t /* codec_id */) @@ -252,4 +254,6 @@ int16_t ACMILBC::SetBitRateSafe(const int32_t rate) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_ilbc.h b/webrtc/modules/audio_coding/main/source/acm_ilbc.h index 50b6df9a9a..bd2495fe31 100644 --- a/webrtc/modules/audio_coding/main/source/acm_ilbc.h +++ b/webrtc/modules/audio_coding/main/source/acm_ilbc.h @@ -19,6 +19,8 @@ struct iLBC_decinst_t_; namespace webrtc { +namespace acm1 { + class ACMILBC : public ACMGenericCodec { public: explicit ACMILBC(int16_t codec_id); @@ -62,6 +64,8 @@ class ACMILBC : public ACMGenericCodec { iLBC_decinst_t_* decoder_inst_ptr_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_isac.cc b/webrtc/modules/audio_coding/main/source/acm_isac.cc index e22d3f61bd..b9316d6d9f 100644 --- a/webrtc/modules/audio_coding/main/source/acm_isac.cc +++ b/webrtc/modules/audio_coding/main/source/acm_isac.cc @@ -28,6 +28,8 @@ namespace webrtc { +namespace acm1 { + // we need this otherwise we cannot use forward declaration // in the header file #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) @@ -896,4 +898,6 @@ int16_t ACMISAC::REDPayloadISAC(const int32_t isac_rate, #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_isac.h b/webrtc/modules/audio_coding/main/source/acm_isac.h index 9588723cb5..20b6c5391b 100644 --- a/webrtc/modules/audio_coding/main/source/acm_isac.h +++ b/webrtc/modules/audio_coding/main/source/acm_isac.h @@ -15,6 +15,8 @@ namespace webrtc { +namespace acm1 { + struct ACMISACInst; enum IsacCodingMode { @@ -129,6 +131,8 @@ class ACMISAC : public ACMGenericCodec { WebRtcACMCodecParams decoder_params_32khz_; }; -} // namespace +} // namespace acm1 + +} // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_isac_macros.h b/webrtc/modules/audio_coding/main/source/acm_isac_macros.h index 6ae4526f57..01e1e44b3e 100644 --- a/webrtc/modules/audio_coding/main/source/acm_isac_macros.h +++ b/webrtc/modules/audio_coding/main/source/acm_isac_macros.h @@ -15,6 +15,8 @@ namespace webrtc { +namespace acm1 { + #ifdef WEBRTC_CODEC_ISAC #define ACM_ISAC_CREATE WebRtcIsac_Create #define ACM_ISAC_FREE WebRtcIsac_Free @@ -67,7 +69,9 @@ namespace webrtc { #define ACM_ISAC_GETDECSAMPRATE ACMISACFixGetDecSampRate // local Impl #endif -} // namespace +} // namespace acm1 + +} // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq.cc b/webrtc/modules/audio_coding/main/source/acm_neteq.cc index 5418d18025..2ade7bf579 100644 --- a/webrtc/modules/audio_coding/main/source/acm_neteq.cc +++ b/webrtc/modules/audio_coding/main/source/acm_neteq.cc @@ -26,6 +26,8 @@ namespace webrtc { +namespace acm1 { + #define RTP_HEADER_SIZE 12 #define NETEQ_INIT_FREQ 8000 #define NETEQ_INIT_FREQ_KHZ (NETEQ_INIT_FREQ/1000) @@ -1140,4 +1142,6 @@ bool ACMNetEQ::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const { return true; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq.h b/webrtc/modules/audio_coding/main/source/acm_neteq.h index 511968b287..e52ddc7957 100644 --- a/webrtc/modules/audio_coding/main/source/acm_neteq.h +++ b/webrtc/modules/audio_coding/main/source/acm_neteq.h @@ -12,8 +12,6 @@ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ #include "webrtc/common_audio/vad/include/webrtc_vad.h" -#include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" #include "webrtc/modules/interface/module_common_types.h" @@ -25,6 +23,8 @@ class CriticalSectionWrapper; class RWLockWrapper; struct CodecInst; +namespace acm1 { + #define MAX_NUM_SLAVE_NETEQ 1 class ACMNetEQ { @@ -392,6 +392,8 @@ class ACMNetEQ { int maximum_delay_ms_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc b/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc index aef0acdde0..8b973ba230 100644 --- a/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc +++ b/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc @@ -24,6 +24,8 @@ namespace webrtc { +namespace acm1 { + class AcmNetEqTest : public ::testing::Test { protected: static const size_t kMaxPayloadLen = 5760; // 60 ms, 48 kHz, 16 bit samples. @@ -146,4 +148,6 @@ TEST_F(AcmNetEqTest, TestZeroLengthWaitingTimesVector) { EXPECT_EQ(-1, stats.medianWaitingTimeMs); } -} // namespace +} // namespace acm1 + +} // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.cc b/webrtc/modules/audio_coding/main/source/acm_opus.cc index 8ea5d51d5b..3a619d04e7 100644 --- a/webrtc/modules/audio_coding/main/source/acm_opus.cc +++ b/webrtc/modules/audio_coding/main/source/acm_opus.cc @@ -23,6 +23,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_OPUS ACMOpus::ACMOpus(int16_t /* codec_id */) @@ -312,4 +314,6 @@ void ACMOpus::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { #endif // WEBRTC_CODEC_OPUS +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.h b/webrtc/modules/audio_coding/main/source/acm_opus.h index fa188a1cb0..1e586ff41a 100644 --- a/webrtc/modules/audio_coding/main/source/acm_opus.h +++ b/webrtc/modules/audio_coding/main/source/acm_opus.h @@ -19,6 +19,8 @@ struct WebRtcOpusDecInst; namespace webrtc { +namespace acm1 { + class ACMOpus : public ACMGenericCodec { public: explicit ACMOpus(int16_t codec_id); @@ -69,6 +71,8 @@ class ACMOpus : public ACMGenericCodec { int channels_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc index 91cb9e03d2..b0032b8607 100644 --- a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc +++ b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc @@ -23,6 +23,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_PCM16 ACMPCM16B::ACMPCM16B(int16_t /* codec_id */) { @@ -244,4 +246,6 @@ void ACMPCM16B::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { } #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_pcm16b.h b/webrtc/modules/audio_coding/main/source/acm_pcm16b.h index 38de343760..a97589b57a 100644 --- a/webrtc/modules/audio_coding/main/source/acm_pcm16b.h +++ b/webrtc/modules/audio_coding/main/source/acm_pcm16b.h @@ -15,6 +15,8 @@ namespace webrtc { +namespace acm1 { + class ACMPCM16B : public ACMGenericCodec { public: explicit ACMPCM16B(int16_t codec_id); @@ -58,6 +60,8 @@ class ACMPCM16B : public ACMGenericCodec { int32_t sampling_freq_hz_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_pcma.cc b/webrtc/modules/audio_coding/main/source/acm_pcma.cc index 83c124922c..c64641771c 100644 --- a/webrtc/modules/audio_coding/main/source/acm_pcma.cc +++ b/webrtc/modules/audio_coding/main/source/acm_pcma.cc @@ -21,6 +21,8 @@ namespace webrtc { +namespace acm1 { + ACMPCMA::ACMPCMA(int16_t codec_id) { codec_id_ = codec_id; } @@ -127,4 +129,6 @@ void ACMPCMA::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { } } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_pcma.h b/webrtc/modules/audio_coding/main/source/acm_pcma.h index 2fc4ea4fe9..cb506eaa6e 100644 --- a/webrtc/modules/audio_coding/main/source/acm_pcma.h +++ b/webrtc/modules/audio_coding/main/source/acm_pcma.h @@ -15,6 +15,8 @@ namespace webrtc { +namespace acm1 { + class ACMPCMA : public ACMGenericCodec { public: explicit ACMPCMA(int16_t codec_id); @@ -56,6 +58,8 @@ class ACMPCMA : public ACMGenericCodec { int32_t* payload_length) OVERRIDE; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc index 61a64ace97..5b6a4575ff 100644 --- a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc +++ b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc @@ -21,6 +21,8 @@ namespace webrtc { +namespace acm1 { + ACMPCMU::ACMPCMU(int16_t codec_id) { codec_id_ = codec_id; } @@ -129,4 +131,6 @@ void ACMPCMU::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { } } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_pcmu.h b/webrtc/modules/audio_coding/main/source/acm_pcmu.h index 309d318704..ea401d59c9 100644 --- a/webrtc/modules/audio_coding/main/source/acm_pcmu.h +++ b/webrtc/modules/audio_coding/main/source/acm_pcmu.h @@ -15,6 +15,8 @@ namespace webrtc { +namespace acm1 { + class ACMPCMU : public ACMGenericCodec { public: explicit ACMPCMU(int16_t codec_id); @@ -56,6 +58,8 @@ class ACMPCMU : public ACMGenericCodec { int32_t* payload_length) OVERRIDE; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_red.cc b/webrtc/modules/audio_coding/main/source/acm_red.cc index 6e7ae9fe70..bc44c7231e 100644 --- a/webrtc/modules/audio_coding/main/source/acm_red.cc +++ b/webrtc/modules/audio_coding/main/source/acm_red.cc @@ -18,6 +18,8 @@ namespace webrtc { +namespace acm1 { + ACMRED::ACMRED(int16_t codec_id) { codec_id_ = codec_id; } @@ -101,4 +103,6 @@ void ACMRED::DestructDecoderSafe() { return; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_red.h b/webrtc/modules/audio_coding/main/source/acm_red.h index c7bad41630..ede18b5218 100644 --- a/webrtc/modules/audio_coding/main/source/acm_red.h +++ b/webrtc/modules/audio_coding/main/source/acm_red.h @@ -15,6 +15,8 @@ namespace webrtc { +namespace acm1 { + class ACMRED : public ACMGenericCodec { public: explicit ACMRED(int16_t codec_id); @@ -53,6 +55,8 @@ class ACMRED : public ACMGenericCodec { virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_resampler.cc b/webrtc/modules/audio_coding/main/source/acm_resampler.cc index 034dbe550f..50ddab1d8b 100644 --- a/webrtc/modules/audio_coding/main/source/acm_resampler.cc +++ b/webrtc/modules/audio_coding/main/source/acm_resampler.cc @@ -17,6 +17,8 @@ namespace webrtc { +namespace acm1 { + ACMResampler::ACMResampler() { } @@ -56,4 +58,6 @@ int16_t ACMResampler::Resample10Msec(const int16_t* in_audio, return out_length / num_audio_channels; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_resampler.h b/webrtc/modules/audio_coding/main/source/acm_resampler.h index c23abb8823..b50e722c44 100644 --- a/webrtc/modules/audio_coding/main/source/acm_resampler.h +++ b/webrtc/modules/audio_coding/main/source/acm_resampler.h @@ -16,6 +16,8 @@ namespace webrtc { +namespace acm1 { + class ACMResampler { public: ACMResampler(); @@ -31,6 +33,8 @@ class ACMResampler { PushResampler resampler_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_ diff --git a/webrtc/modules/audio_coding/main/source/acm_speex.cc b/webrtc/modules/audio_coding/main/source/acm_speex.cc index ce205266a7..5752693472 100644 --- a/webrtc/modules/audio_coding/main/source/acm_speex.cc +++ b/webrtc/modules/audio_coding/main/source/acm_speex.cc @@ -25,6 +25,8 @@ namespace webrtc { +namespace acm1 { + #ifndef WEBRTC_CODEC_SPEEX ACMSPEEX::ACMSPEEX(int16_t /* codec_id */) : encoder_inst_ptr_(NULL), @@ -464,4 +466,6 @@ int16_t ACMSPEEX::SetComplMode(int16_t mode) { #endif +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/acm_speex.h b/webrtc/modules/audio_coding/main/source/acm_speex.h index 0f62ea34af..762aea8d9c 100644 --- a/webrtc/modules/audio_coding/main/source/acm_speex.h +++ b/webrtc/modules/audio_coding/main/source/acm_speex.h @@ -19,6 +19,8 @@ struct SPEEX_decinst_t_; namespace webrtc { +namespace acm1 { + class ACMSPEEX : public ACMGenericCodec { public: explicit ACMSPEEX(int16_t codec_id); @@ -77,6 +79,8 @@ class ACMSPEEX : public ACMGenericCodec { uint16_t samples_in_20ms_audio_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_ diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc index 4ac40b77df..9461a1f184 100644 --- a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc +++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc @@ -21,31 +21,31 @@ namespace webrtc { // Create module AudioCodingModule* AudioCodingModule::Create(const int32_t id) { - return new AudioCodingModuleImpl(id, Clock::GetRealTimeClock()); + return new acm1::AudioCodingModuleImpl(id, Clock::GetRealTimeClock()); } // Used for testing by inserting a simulated clock. ACM will not destroy the // injected |clock| the client has to take care of that. AudioCodingModule* AudioCodingModule::Create(const int32_t id, Clock* clock) { - return new AudioCodingModuleImpl(id, clock); + return new acm1::AudioCodingModuleImpl(id, clock); } // Destroy module void AudioCodingModule::Destroy(AudioCodingModule* module) { - delete static_cast(module); + delete static_cast(module); } // Get number of supported codecs uint8_t AudioCodingModule::NumberOfCodecs() { - return static_cast(ACMCodecDB::kNumCodecs); + return static_cast(acm1::ACMCodecDB::kNumCodecs); } // Get supported codec param with id int32_t AudioCodingModule::Codec(uint8_t list_id, CodecInst* codec) { // Get the codec settings for the codec with the given list ID - return ACMCodecDB::Codec(list_id, codec); + return acm1::ACMCodecDB::Codec(list_id, codec); } // Get supported codec Param with name, frequency and number of channels. @@ -55,7 +55,8 @@ int32_t AudioCodingModule::Codec(const char* payload_name, int codec_id; // Get the id of the codec from the database. - codec_id = ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels); + codec_id = acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, + channels); if (codec_id < 0) { // We couldn't find a matching codec, set the parameters to unacceptable // values and return. @@ -68,7 +69,7 @@ int32_t AudioCodingModule::Codec(const char* payload_name, } // Get default codec settings. - ACMCodecDB::Codec(codec_id, codec); + acm1::ACMCodecDB::Codec(codec_id, codec); // Keep the number of channels from the function call. For most codecs it // will be the same value as in default codec settings, but not for all. @@ -80,14 +81,14 @@ int32_t AudioCodingModule::Codec(const char* payload_name, // Get supported codec Index with name, frequency and number of channels. int32_t AudioCodingModule::Codec(const char* payload_name, int sampling_freq_hz, int channels) { - return ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels); + return acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels); } // Checks the validity of the parameters of the given codec bool AudioCodingModule::IsCodecValid(const CodecInst& codec) { int mirror_id; - int codec_number = ACMCodecDB::CodecNumber(&codec, &mirror_id); + int codec_number = acm1::ACMCodecDB::CodecNumber(&codec, &mirror_id); if (codec_number < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, @@ -99,8 +100,8 @@ bool AudioCodingModule::IsCodecValid(const CodecInst& codec) { } AudioCodingModule* AudioCodingModuleFactory::Create(int id) const { - return new AudioCodingModuleImpl(static_cast(id), - Clock::GetRealTimeClock()); + return new acm1::AudioCodingModuleImpl(static_cast(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 b136d84ca7..93b21e68dc 100644 --- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc @@ -32,6 +32,8 @@ namespace webrtc { +namespace acm1 { + enum { kACMToneEnd = 999 }; @@ -3115,4 +3117,6 @@ void AudioCodingModuleImpl::DisableNack() { nack_enabled_ = false; } +} // namespace acm1 + } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h index f58e3e5ed3..64afe4f8e0 100644 --- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h +++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h @@ -15,6 +15,7 @@ #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" #include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" #include "webrtc/modules/audio_coding/main/source/acm_neteq.h" #include "webrtc/modules/audio_coding/main/source/acm_resampler.h" @@ -22,11 +23,16 @@ namespace webrtc { -class ACMDTMFDetection; -class ACMGenericCodec; class CriticalSectionWrapper; class RWLockWrapper; class Clock; + +namespace acm1 { + +struct WebRtcACMAudioBuff; +struct WebRtcACMCodecParams; +class ACMDTMFDetection; +class ACMGenericCodec; class Nack; class AudioCodingModuleImpl : public AudioCodingModule { @@ -88,8 +94,7 @@ class AudioCodingModuleImpl : public AudioCodingModule { // Register a transport callback which will be // called to deliver the encoded buffers. - int32_t RegisterTransportCallback( - AudioPacketizationCallback* transport); + int32_t RegisterTransportCallback(AudioPacketizationCallback* transport); // Used by the module to deliver messages to the codec module/application // AVT(DTMF). @@ -125,8 +130,7 @@ class AudioCodingModuleImpl : public AudioCodingModule { bool enable_vad = false, ACMVADMode mode = VADNormal); - int32_t VAD(bool* dtx_enabled, bool* vad_enabled, - ACMVADMode* mode) const; + int32_t VAD(bool* dtx_enabled, bool* vad_enabled, ACMVADMode* mode) const; int32_t RegisterVADCallback(ACMVADCallback* vad_callback); @@ -454,6 +458,8 @@ class AudioCodingModuleImpl : public AudioCodingModule { bool nack_enabled_; }; +} // namespace acm1 + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_ diff --git a/webrtc/modules/audio_coding/main/source/nack.cc b/webrtc/modules/audio_coding/main/source/nack.cc index ec6cb3d12c..4ca260ddc2 100644 --- a/webrtc/modules/audio_coding/main/source/nack.cc +++ b/webrtc/modules/audio_coding/main/source/nack.cc @@ -19,6 +19,8 @@ namespace webrtc { +namespace acm1 { + namespace { const int kDefaultSampleRateKhz = 48; @@ -222,4 +224,6 @@ std::vector Nack::GetNackList(int round_trip_time_ms) const { return sequence_numbers; } -} // webrtc +} // namespace acm1 + +} // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/source/nack.h b/webrtc/modules/audio_coding/main/source/nack.h index e047c28245..9cea15d1a2 100644 --- a/webrtc/modules/audio_coding/main/source/nack.h +++ b/webrtc/modules/audio_coding/main/source/nack.h @@ -49,6 +49,8 @@ // namespace webrtc { +namespace acm1 { + class Nack { public: // A limit for the size of the NACK list. @@ -204,6 +206,8 @@ class Nack { size_t max_nack_list_size_; }; -} // webrtc +} // namespace acm1 + +} // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_NACK_H_ diff --git a/webrtc/modules/audio_coding/main/source/nack_unittest.cc b/webrtc/modules/audio_coding/main/source/nack_unittest.cc index ba92f0e764..811aca4fc8 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 #include +#include #include "gtest/gtest.h" #include "webrtc/typedefs.h" @@ -21,6 +22,8 @@ namespace webrtc { +namespace acm1 { + namespace { const int kNackThreshold = 3; @@ -479,4 +482,6 @@ TEST(NackTest, RoudTripTimeIsApplied) { EXPECT_EQ(5, nack_list[1]); } -} // webrtc +} // namespace acm1 + +} // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/main/test/opus_test.h index ca0da9ee15..49b98ea862 100644 --- a/webrtc/modules/audio_coding/main/test/opus_test.h +++ b/webrtc/modules/audio_coding/main/test/opus_test.h @@ -29,6 +29,7 @@ class OpusTest : public ACMTest { ~OpusTest(); void Perform(); + private: void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length, int percent_loss = 0); @@ -44,7 +45,7 @@ class OpusTest : public ACMTest { int counter_; uint8_t payload_type_; int rtp_timestamp_; - ACMResampler resampler_; + acm1::ACMResampler resampler_; WebRtcOpusEncInst* opus_mono_encoder_; WebRtcOpusEncInst* opus_stereo_encoder_; WebRtcOpusDecInst* opus_mono_decoder_;