1999-12-05 01:40:43 +00:00
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#
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2012-10-14 21:56:13 +00:00
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# Asterisk -- An open source telephony toolkit.
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2017-12-22 14:23:22 +00:00
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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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2012-09-25 17:02:21 +00:00
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SUB_GSM := gsm
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SUB_ILBC := ilbc
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LIBILBC := $(SUB_ILBC)/libilbc.a
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SUB_LPC10 := lpc10
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LIBLPC10 := $(SUB_LPC10)/liblpc10.a
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SUB_DIRS := \
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$(SUB_GSM) \
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$(SUB_ILBC) \
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$(SUB_LPC10) \
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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 $(ASTTOPDIR)/Makefile.moddir_rules
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1999-12-05 01:40:43 +00:00
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2012-09-25 17:02:21 +00:00
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2007-01-04 18:19:55 +00:00
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ifneq ($(GSM_INTERNAL),no)
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2012-09-25 17:02:21 +00:00
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GSM_INCLUDE := -I$(SUB_GSM)/inc
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2006-06-24 23:12:22 +00:00
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2017-03-24 12:43:05 +00:00
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codec_gsm.so: $(SUB_GSM)/lib/libgsm.a
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2012-07-23 21:27:56 +00:00
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endif
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2012-09-25 17:02:21 +00:00
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# Don't run the implicit rules for this target.
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$(SUB_GSM)/lib/libgsm.a: $(SUB_GSM) ;
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$(SUB_GSM):
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@mkdir -p $(SUB_GSM)/lib
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@$(MAKE) -C $(SUB_GSM) lib/libgsm.a
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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: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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2012-09-25 17:02:21 +00:00
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for dir in $(SUB_DIRS); do \
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$(MAKE) -C $$dir clean; \
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done
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1999-12-05 01:40:43 +00:00
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2012-09-25 17:02:21 +00:00
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.PHONY: $(SUB_DIRS)
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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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2017-03-24 12:43:05 +00:00
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codec_lpc10.so: $(LIBLPC10)
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2006-06-24 23:12:22 +00:00
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2012-09-25 17:02:21 +00:00
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# Don't run the implicit rules for this target.
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$(LIBLPC10): $(SUB_LPC10) ;
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$(SUB_LPC10):
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@$(MAKE) -C $(SUB_LPC10) all
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ifneq ($(ILBC_INTERNAL),no)
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2017-03-24 12:43:05 +00:00
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codec_ilbc.so: $(LIBILBC)
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2012-09-25 17:02:21 +00:00
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else
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ILBC_INCLUDE += -DILBC_WEBRTC
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endif
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# Don't run the implicit rules for this target.
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$(LIBILBC): $(SUB_ILBC) ;
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$(SUB_ILBC):
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@$(MAKE) -C $(SUB_ILBC) all _ASTCFLAGS="$(filter-out -Wmissing-prototypes -Wmissing-declarations -Wshadow,$(_ASTCFLAGS)) $(AST_NO_STRICT_OVERFLOW)"
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2003-04-15 04:36:52 +00:00
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
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2012-09-25 17:02:21 +00:00
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2015-04-13 10:28:32 +00:00
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$(call MOD_ADD_C,codec_g722,g722/g722_encode.c g722/g722_decode.c)
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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: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
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2012-09-25 17:02:21 +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: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
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ifeq ($(BUILD_CPU),x86_64)
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SPEEX_RESAMPLE_CFLAGS:=-fPIC
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else
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SPEEX_RESAMPLE_CFLAGS:=
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endif
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2015-04-13 10:28:32 +00:00
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$(call MOD_ADD_C,codec_resample,speex/resample.c)
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2019-09-08 15:38:57 +00:00
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codec_resample.o: _ASTCFLAGS+=-DOUTSIDE_SPEEX
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2019-08-23 20:14:36 +00:00
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speex/resample.o: _ASTCFLAGS+=$(SPEEX_RESAMPLE_CFLAGS) -DOUTSIDE_SPEEX -DEXPORT=
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