Hello,
I patched manually the version 1.4.25.1 recently, and it's working great.
Older version of Asterisk 1.4 are not always well supported by last
versions of asterisk-video.
For exemple, I had 1.4.15.1 patched all right but some news functions
needed by app_h324m were missing causing some compilation warnings.
app_h324m compiled ok and loaded fine too but h324m_gw() didn't worked
at all and made asterisk crash.. while h324m_loopback worked fine...
So I upgraded asterisk to the last version (1.4.25.1) and patched it
manually.
Look at my attached patch for this version.
Regards
Piotr Stankiewicz a écrit :
W dniu 10 lipca 2009 10:23 użytkownik Konrad Różycki
<[email protected]> napisał:
Hello,
Is there any AMR codec for Asterisk 1.4.24.1. I trying to use AMR
from SVN (http://sip.fontventa.com/svn/asterisk/amr) but without
sucess:
patching file build_tools/embed_modules.xml
patching file include/asterisk/frame.h
Hunk #1 FAILED at 242.
1 out of 1 hunk FAILED -- saving rejects to file
include/asterisk/frame.h.rej
patching file main/channel.c
Hunk #1 succeeded at 677 (offset -18 lines).
patching file main/translate.c
Hunk #1 succeeded at 523 (offset 31 lines).
patching file main/rtp.c
Hunk #1 succeeded at 1362 (offset 21 lines).
patching file main/frame.c
Hunk #1 succeeded at 119 (offset 2 lines).
Hunk #2 succeeded at 663 (offset 32 lines).
Hunk #3 succeeded at 677 (offset 2 lines).
Hunk #4 succeeded at 1513 (offset 51 lines).
Hunk #5 succeeded at 1531 (offset 2 lines).
patching file codecs/Makefile
Hunk #1 succeeded at 31 (offset 6 lines).
Hunk #3 succeeded at 61 with fuzz 1 (offset 6 lines).
I looking also for proper user information layer 1 handling patch for
Asterisk 1.4.24.1, maybe someone prepared it for that version of * ?
(need to pass throgh video call from 3G network to video platform).
Hi Konrad,
Try earlier version of asterisk. Mine is 1.4.19.2 and everything works
perfectly.
Regards,
Void.
Best Rgerards
--
Konrad Rozycki
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Julien LEBOEUF
Technicien Réseau
NEOCOM Multimedia
70, Boulevard Louis Blanc
85000 La Roche sur Yon
Tél. + 33 (0)2 53 59 30 00 extension 50
Fax. + 33 (0)2 51 62 00 51
Skype : julien.leboeuf
[email protected]
www.neocom.fr
diff -cr asterisk-1.4.25.1-orig/build_tools/embed_modules.xml
asterisk-1.4.25.1/build_tools/embed_modules.xml
*** asterisk-1.4.25.1-orig/build_tools/embed_modules.xml 2009-07-08
16:04:51.000000000 +0200
--- asterisk-1.4.25.1/build_tools/embed_modules.xml 2009-07-08
16:12:49.000000000 +0200
***************
*** 8,14 ****
<member name="channels" displayname="Channels"
remove_on_change="channels/*.o channels/misdn/*.o">
<depend>gnu_ld</depend>
</member>
! <member name="codecs" displayname="Coders/Decoders"
remove_on_change="codecs/*.o codecs/gsm/src/*.o codecs/ilbc/*.o
codecs/lpc10/*.o codecs/gsm/lib/libgsm.a codecs/lpc10/liblpc10.a
codecs/ilbc/libilbc.a">
<depend>gnu_ld</depend>
</member>
<member name="formats" displayname="File Formats"
remove_on_change="formats/*.o">
--- 8,14 ----
<member name="channels" displayname="Channels"
remove_on_change="channels/*.o channels/misdn/*.o">
<depend>gnu_ld</depend>
</member>
! <member name="codecs" displayname="Coders/Decoders"
remove_on_change="codecs/*.o codecs/gsm/src/*.o codecs/ilbc/*.o
codecs/lpc10/*.o codecs/gsm/lib/libgsm.a codecs/lpc10/liblpc10.a
codecs/ilbc/libilbc.a codecs/amr/libamr.a">
<depend>gnu_ld</depend>
</member>
<member name="formats" displayname="File Formats"
remove_on_change="formats/*.o">
diff -cr asterisk-1.4.25.1-orig/codecs/Makefile
asterisk-1.4.25.1/codecs/Makefile
*** asterisk-1.4.25.1-orig/codecs/Makefile 2009-07-08 16:04:51.000000000
+0200
--- asterisk-1.4.25.1/codecs/Makefile 2009-07-08 17:11:57.000000000 +0200
***************
*** 31,36 ****
--- 31,37 ----
LIBILBC:=ilbc/libilbc.a
LIBLPC10:=lpc10/liblpc10.a
+ LIBAMR:=amr/libamr.a
all: _all
***************
*** 45,50 ****
--- 46,52 ----
$(MAKE) -C gsm clean
$(MAKE) -C lpc10 clean
$(MAKE) -C ilbc clean
+ $(MAKE) -C amr clean
gsm/lib/libgsm.a:
@mkdir -p gsm/lib
***************
*** 59,61 ****
--- 61,69 ----
@$(MAKE) -C ilbc all ASTCFLAGS="$(filter-out -Wmissing-prototypes
-Wmissing-declarations,$(ASTCFLAGS)) $(AST_NO_STRICT_OVERFLOW)"
$(if $(filter codec_ilbc,$(EMBEDDED_MODS)),modules.link,codec_ilbc.so):
$(LIBILBC)
+
+ $(LIBAMR):
+ @$(MAKE) -C amr all
+
+ $(if $(filter codec_amr,$(EMBEDDED_MODS)),modules.link,codec_amr.so):
$(LIBAMR)
+
diff -cr asterisk-1.4.25.1-orig/include/asterisk/frame.h
asterisk-1.4.25.1/include/asterisk/frame.h
*** asterisk-1.4.25.1-orig/include/asterisk/frame.h 2009-07-08
16:04:51.000000000 +0200
--- asterisk-1.4.25.1/include/asterisk/frame.h 2009-07-08 16:14:42.000000000
+0200
***************
*** 260,267 ****
#define AST_FORMAT_G726 (1 << 11)
/*! G.722 */
#define AST_FORMAT_G722 (1 << 12)
/*! Unsupported audio bits */
! #define AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14) | (1 << 15))
/*! Maximum audio format */
#define AST_FORMAT_MAX_AUDIO (1 << 15)
/*! Maximum audio mask */
--- 260,269 ----
#define AST_FORMAT_G726 (1 << 11)
/*! G.722 */
#define AST_FORMAT_G722 (1 << 12)
+ /*! AMR-NB */
+ #define AST_FORMAT_AMRNB (1 << 13)
/*! Unsupported audio bits */
! #define AST_FORMAT_AUDIO_UNDEFINED ((1 << 14) | (1 << 15))
/*! Maximum audio format */
#define AST_FORMAT_MAX_AUDIO (1 << 15)
/*! Maximum audio mask */
diff -cr asterisk-1.4.25.1-orig/main/channel.c asterisk-1.4.25.1/main/channel.c
*** asterisk-1.4.25.1-orig/main/channel.c 2009-07-08 16:04:51.000000000
+0200
--- asterisk-1.4.25.1/main/channel.c 2009-07-08 16:24:24.000000000 +0200
***************
*** 677,682 ****
--- 677,684 ----
/*! Okay, we're down to vocoders now, so pick GSM because it's
small and easier to
translate and sounds pretty good */
AST_FORMAT_GSM,
+ /*! Try AMR */
+ AST_FORMAT_AMRNB,
/*! iLBC is not too bad */
AST_FORMAT_ILBC,
/*! Speex is free, but computationally more expensive than GSM
*/
diff -cr asterisk-1.4.25.1-orig/main/frame.c asterisk-1.4.25.1/main/frame.c
*** asterisk-1.4.25.1-orig/main/frame.c 2009-07-08 16:04:51.000000000 +0200
--- asterisk-1.4.25.1/main/frame.c 2009-07-08 16:59:18.000000000 +0200
***************
*** 119,125 ****
{ 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 },
/*!< 11: codec_ilbc.c */ /* inc=30ms - workaround */
{ 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 300, 10,
20 }, /*!< 12: codec_g726.c */
{ 1, AST_FORMAT_G722, "g722", "G722"},
/*!< 13 */
! { 0, 0, "nothing", "undefined" },
{ 0, 0, "nothing", "undefined" },
{ 0, 0, "nothing", "undefined" },
{ 0, 0, "nothing", "undefined" },
--- 119,125 ----
{ 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 },
/*!< 11: codec_ilbc.c */ /* inc=30ms - workaround */
{ 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 300, 10,
20 }, /*!< 12: codec_g726.c */
{ 1, AST_FORMAT_G722, "g722", "G722"},
/*!< 13 */
! { 1, AST_FORMAT_AMRNB, "amr", "AMR NB", 32, 20, 300, 0, 20},
/*!< codec_amr.c */
{ 0, 0, "nothing", "undefined" },
{ 0, 0, "nothing", "undefined" },
{ 0, 0, "nothing", "undefined" },
***************
*** 665,671 ****
ast_cli(fd,
"--------------------------------------------------------------------------------\n");
if ((argc == 2) || (!strcasecmp(argv[1],"audio"))) {
found = 1;
! for (i=0;i<13;i++) {
snprintf(hex,25,"(0x%x)",1<<i);
ast_cli(fd, "%11u (1 << %2d) %10s audio %8s
(%s)\n",1 << i,i,hex,ast_getformatname(1<<i),ast_codec2str(1<<i));
}
--- 665,671 ----
ast_cli(fd,
"--------------------------------------------------------------------------------\n");
if ((argc == 2) || (!strcasecmp(argv[1],"audio"))) {
found = 1;
! for (i=0;i<14;i++) {
snprintf(hex,25,"(0x%x)",1<<i);
ast_cli(fd, "%11u (1 << %2d) %10s audio %8s
(%s)\n",1 << i,i,hex,ast_getformatname(1<<i),ast_codec2str(1<<i));
}
***************
*** 709,715 ****
ast_cli(fd,
"--------------------------------------------------------------------------------\n");
if ((argc == 3) || (!strcasecmp(argv[3],"audio"))) {
found = 1;
! for (i=0;i<13;i++) {
snprintf(hex,25,"(0x%x)",1<<i);
ast_cli(fd, "%11u (1 << %2d) %10s audio %8s
(%s)\n",1 << i,i,hex,ast_getformatname(1<<i),ast_codec2str(1<<i));
}
--- 709,715 ----
ast_cli(fd,
"--------------------------------------------------------------------------------\n");
if ((argc == 3) || (!strcasecmp(argv[3],"audio"))) {
found = 1;
! for (i=0;i<14;i++) {
snprintf(hex,25,"(0x%x)",1<<i);
ast_cli(fd, "%11u (1 << %2d) %10s audio %8s
(%s)\n",1 << i,i,hex,ast_getformatname(1<<i),ast_codec2str(1<<i));
}
***************
*** 1515,1520 ****
--- 1515,1550 ----
return cnt;
}
+ static int amr_samples(unsigned char *data, int datalen)
+ {
+ short block_size[16]={ 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0,
0, 0, 0 };
+ int samples = 0;
+
+ /* we asume RTP octed aligned format (RFC 4867 section 4.4)
+ skip first octed (CMR) */
+ datalen--;
+ data++;
+
+ while (datalen > 0) {
+ unsigned dec_mode = (data[0]>>3) & 0x000f;
+ unsigned psize = block_size[dec_mode];
+
+ if (psize)
+ samples += 160;
+
+ if ( !(data[0]>>7) ) {
+ /* this was the last ToC entry */
+ break;
+ }
+
+ data++; /* Skip over the block. */
+ datalen --;
+ /* ast_verbose("Got AMR frame size %d inside ast_frame\n",
psize); */
+ }
+ /* todo: how can we signal an error while decoding the frame structure?
*/
+ return samples;
+ }
+
int ast_codec_get_samples(struct ast_frame *f)
{
int samples=0;
***************
*** 1552,1557 ****
--- 1582,1590 ----
case AST_FORMAT_G726_AAL2:
samples = f->datalen * 2;
break;
+ case AST_FORMAT_AMRNB:
+ samples = amr_samples(f->data, f->datalen);
+ break;
default:
ast_log(LOG_WARNING, "Unable to calculate samples for format
%s\n", ast_getformatname(f->subclass));
}
diff -cr asterisk-1.4.25.1-orig/main/rtp.c asterisk-1.4.25.1/main/rtp.c
*** asterisk-1.4.25.1-orig/main/rtp.c 2009-07-08 16:04:51.000000000 +0200
--- asterisk-1.4.25.1/main/rtp.c 2009-07-08 16:53:53.000000000 +0200
***************
*** 1397,1402 ****
--- 1397,1403 ----
{{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
{{1, AST_FORMAT_G722}, "audio", "G722"},
{{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32"},
+ {{1, AST_FORMAT_AMRNB}, "audio", "AMR"},
{{0, AST_RTP_DTMF}, "audio", "telephone-event"},
{{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
{{0, AST_RTP_CN}, "audio", "CN"},
***************
*** 1434,1439 ****
--- 1435,1441 ----
[31] = {1, AST_FORMAT_H261},
[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
+ [96] = {1, AST_FORMAT_AMRNB},
[97] = {1, AST_FORMAT_ILBC},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
diff -cr asterisk-1.4.25.1-orig/main/translate.c
asterisk-1.4.25.1/main/translate.c
*** asterisk-1.4.25.1-orig/main/translate.c 2009-07-08 16:04:51.000000000
+0200
--- asterisk-1.4.25.1/main/translate.c 2009-07-08 16:52:32.000000000 +0200
***************
*** 523,529 ****
/*! \brief CLI "show translation" command handler */
static int show_translation_deprecated(int fd, int argc, char *argv[])
{
! #define SHOW_TRANS 13
int x, y, z;
int curlen = 0, longest = 0;
--- 523,529 ----
/*! \brief CLI "show translation" command handler */
static int show_translation_deprecated(int fd, int argc, char *argv[])
{
! #define SHOW_TRANS 14
int x, y, z;
int curlen = 0, longest = 0;
diff -cr asterisk-1.4.25.1-orig/menuselect-tree
asterisk-1.4.25.1/menuselect-tree
*** asterisk-1.4.25.1-orig/menuselect-tree 2009-07-08 16:04:51.000000000
+0200
--- asterisk-1.4.25.1/menuselect-tree 2009-07-08 17:09:32.000000000 +0200
***************
*** 492,498 ****
<member name="channels" displayname="Channels"
remove_on_change="channels/*.o channels/misdn/*.o">
<depend>gnu_ld</depend>
</member>
! <member name="codecs" displayname="Coders/Decoders"
remove_on_change="codecs/*.o codecs/gsm/src/*.o codecs/ilbc/*.o
codecs/lpc10/*.o codecs/gsm/lib/libgsm.a codecs/lpc10/liblpc10.a
codecs/ilbc/libilbc.a">
<depend>gnu_ld</depend>
</member>
<member name="formats" displayname="File Formats"
remove_on_change="formats/*.o">
--- 492,498 ----
<member name="channels" displayname="Channels"
remove_on_change="channels/*.o channels/misdn/*.o">
<depend>gnu_ld</depend>
</member>
! <member name="codecs" displayname="Coders/Decoders"
remove_on_change="codecs/*.o codecs/gsm/src/*.o codecs/ilbc/*.o
codecs/lpc10/*.o codecs/gsm/lib/libgsm.a codecs/lpc10/liblpc10.a
codecs/ilbc/libilbc.a codecs/amr/libamr.a">
<depend>gnu_ld</depend>
</member>
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