1999-12-05 01:40:43 +00:00
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#
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# Asterisk -- A telephony toolkit for Linux.
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#
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2005-06-20 17:26:08 +00:00
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# Makefile for codec modules
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1999-12-05 01:40:43 +00:00
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#
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2006-02-11 17:41:36 +00:00
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# Copyright (C) 1999-2006, Digium, Inc.
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1999-12-05 01:40:43 +00:00
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#
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2005-06-20 17:26:08 +00:00
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# Mark Spencer <markster@digium.com>
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1999-12-05 01:40:43 +00:00
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#
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# This program is free software, distributed under the terms of
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# the GNU General Public License
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#
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2007-12-09 21:29:37 +00:00
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-include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
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2006-06-07 16:03:31 +00:00
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2007-12-17 07:25:35 +00:00
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MODULE_PREFIX=codec
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MENUSELECT_CATEGORY=CODECS
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MENUSELECT_DESCRIPTION=Codec Translators
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2005-08-30 02:54:02 +00:00
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2006-06-29 07:50:01 +00:00
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LIBILBC:=ilbc/libilbc.a
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2006-06-24 23:12:22 +00:00
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LIBLPC10:=lpc10/liblpc10.a
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2005-08-30 02:54:02 +00:00
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2006-06-24 23:12:22 +00:00
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all: _all
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2006-06-05 20:46:27 +00:00
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2006-08-21 02:11:39 +00:00
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include $(ASTTOPDIR)/Makefile.moddir_rules
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1999-12-05 01:40:43 +00:00
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2007-01-04 18:19:55 +00:00
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ifneq ($(GSM_INTERNAL),no)
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2006-08-21 02:11:39 +00:00
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GSM_INCLUDE:=-Igsm/inc
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2006-09-29 22:48:43 +00:00
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$(if $(filter codec_gsm,$(EMBEDDED_MODS)),modules.link,codec_gsm.so): gsm/lib/libgsm.a
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2006-06-24 23:12:22 +00:00
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endif
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22 23:04:49 +00:00
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2006-06-24 23:12:22 +00:00
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clean::
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2003-04-23 16:23:12 +00:00
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$(MAKE) -C gsm clean
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$(MAKE) -C lpc10 clean
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$(MAKE) -C ilbc clean
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2009-05-15 17:37:12 +00:00
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rm -f g722/*.[oa]
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22 23:04:49 +00:00
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rm -f speex/*.[oa]
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1999-12-05 01:40:43 +00:00
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2006-04-24 17:11:45 +00:00
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gsm/lib/libgsm.a:
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2006-07-06 23:18:45 +00:00
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@mkdir -p gsm/lib
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2006-09-29 18:54:21 +00:00
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@$(MAKE) -C gsm lib/libgsm.a
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1999-12-05 01:40:43 +00:00
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2000-01-05 17:22:42 +00:00
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$(LIBLPC10):
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2006-09-29 18:54:21 +00:00
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@$(MAKE) -C lpc10 all
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2000-01-05 17:22:42 +00:00
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2006-09-29 22:48:43 +00:00
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$(if $(filter codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10)
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2006-06-24 23:12:22 +00:00
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2003-04-15 04:36:52 +00:00
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$(LIBILBC):
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2009-07-21 13:28:04 +00:00
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@$(MAKE) -C ilbc all _ASTCFLAGS="$(filter-out -Wmissing-prototypes -Wmissing-declarations -Wshadow,$(_ASTCFLAGS)) $(AST_NO_STRICT_OVERFLOW)"
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2003-04-15 04:36:52 +00:00
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22 23:04:49 +00:00
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2006-09-29 22:48:43 +00:00
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$(if $(filter codec_ilbc,$(EMBEDDED_MODS)),modules.link,codec_ilbc.so): $(LIBILBC)
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2006-12-21 00:08:21 +00:00
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2009-05-15 17:37:12 +00:00
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$(if $(filter codec_g722,$(EMBEDDED_MODS)),modules.link,codec_g722.so): g722/g722_encode.o g722/g722_decode.o
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2010-03-23 14:22:27 +00:00
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g722/g722_encode.o g722/g722_decode.o: _ASTCFLAGS+=$(call MOD_ASTCFLAGS,codec_g722)
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22 23:04:49 +00:00
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ifeq ($(BUILD_CPU),x86_64)
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SPEEX_RESAMPLE_CFLAGS:=-fPIC
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else
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SPEEX_RESAMPLE_CFLAGS:=
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endif
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$(if $(filter codec_resample,$(EMBEDDED_MODS)),modules.link,codec_resample.so): speex/resample.o
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speex/resample.o: _ASTCFLAGS+=$(call MOD_ASTCFLAGS,codec_resample) $(SPEEX_RESAMPLE_CFLAGS)
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