[svn-commits] kpfleming: trunk r370407 - in /trunk: ./ build_tools/ codecs/ include/asterisk/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jul 23 16:28:01 CDT 2012
Author: kpfleming
Date: Mon Jul 23 16:27:56 2012
New Revision: 370407
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370407
Log:
Enable usage of system-provided iLBC library.
The WebRTC version of the iLBC codec is now package as a library and is
available on some platforms. This patch allows codec_ilbc to be built against
that library if it is present.
Review: https://reviewboard.asterisk.org/r/1964/
Modified:
trunk/CHANGES
trunk/build_tools/menuselect-deps.in
trunk/codecs/Makefile
trunk/codecs/codec_ilbc.c
trunk/configure
trunk/configure.ac
trunk/include/asterisk/autoconfig.h.in
trunk/makeopts.in
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=370407&r1=370406&r2=370407
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Mon Jul 23 16:27:56 2012
@@ -78,7 +78,7 @@
-------------------
* Added support for IPv6.
- * Add interrupt ('I') command to ExternalIVR. Sending this command from an
+ * Add interrupt ('I') command to ExternalIVR. Sending this command from an
external process will cause the current playlist to be cleared, including
stopping any audio file that is currently playing. This is useful when you
want to interrupt audio playback only when specific DTMF is entered by the
@@ -254,7 +254,7 @@
* Added NAT support for RTP. Setting in config is 'nat', which can be set
globally and overriden on a peer by peer basis.
- * Direct media functionality has been added. Options in config are:
+ * Direct media functionality has been added. Options in config are:
directmedia (directrtp) and directrtpsetup (earlydirect)
* ChannelUpdate events now contain a CallRef header.
@@ -354,16 +354,16 @@
* Added global 'debug' option, that enables debug in channel driver
* Added ability to translate on-screen menu in multiple languages. Tested on
- Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
- ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
+ Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
+ ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
menu of phone
* In addition to English added French and Russian languages for on-screen menus
- * Reworked dialing number input: added dialing by timeout, immediate dial on
+ * Reworked dialing number input: added dialing by timeout, immediate dial on
on dialplan compare, phone number length now not limited by screen size
- * Added ability to pickup a call using features.conf defined value and
+ * Added ability to pickup a call using features.conf defined value and
on-screen key
@@ -421,7 +421,7 @@
returned if auto_force_rport is not enabled.
* Hangup now can take a regular expression as the Channel option. If you want
- to hangup multiple channels, use /regex/ as the Channel option. Existing
+ to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
@@ -509,6 +509,8 @@
and CELT. You are able to set up a call and have attribute information pass.
This should help considerably with video calls.
+ * The iLBC codec can now use a system-provided iLBC library if one is installed,
+ just like the GSM codec.
Logging
-------------------
@@ -545,7 +547,7 @@
* Channel variable PARKER is now set when comebacktoorigin is disabled in
a parking lot.
- * Channel variable PARKEDCALL is now set with the name of the parking lot
+ * Channel variable PARKEDCALL is now set with the name of the parking lot
when a timeout occurs.
@@ -567,7 +569,7 @@
Calendars
-------------------
- * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
+ * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
CALENDAR_WRITE has completed successfully.
@@ -711,7 +713,7 @@
mixing audio at sample rates ranging from 8khz-96khz.
* CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
and bridge profiles on a channel.
- * CONFBRIDGE_INFO dialplan function capable of retrieving information
+ * CONFBRIDGE_INFO dialplan function capable of retrieving information
about a conference such as locked status and number of parties, admins,
and marked users.
* Addition of video_mode option in confbridge.conf for adding video support
@@ -759,7 +761,7 @@
MixMonitor
--------------------------
- * Added two new options, r and t with file name arguments to record
+ * Added two new options, r and t with file name arguments to record
single direction (unmixed) audio recording separate from the bidirectional
(mixed) recording. The mixed file name argument is optional now as long
as at least one recording option is used.
@@ -1023,7 +1025,7 @@
* Added 'y' option to app_record. This option enables a mode where any DTMF digit
received will terminate recording.
* Voicemail now supports per mailbox settings for folders when using IMAP storage.
- Previously the folder could only be set per context, but has now been extended
+ Previously the folder could only be set per context, but has now been extended
using the imapfolder option.
* Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
* Voicemail now allows the pager date format to be specified separately from the
@@ -1274,7 +1276,7 @@
to eliminate tromboned calls. A tromboned call goes out an interface and comes
back into the same interface. Tromboned calls happen because of call routing,
call deflection, call forwarding, and call transfer.
- * Added the ability to send and receive ETSI Advice-Of-Charge messages.
+ * Added the ability to send and receive ETSI Advice-Of-Charge messages.
* Added the ability to support call waiting calls. (The SETUP has no B channel
assigned.)
* Added Malicious Call ID (MCID) event to the AMI call event class.
@@ -1298,7 +1300,7 @@
* The configuration file manager.conf now supports a channelvars option, which
specifies a list of channel variables to include in each channel-oriented
event.
- * The redirect command now has new parameters ExtraContext, ExtraExtension,
+ * The redirect command now has new parameters ExtraContext, ExtraExtension,
and ExtraPriority to allow redirecting the second channel to a different
location than the first.
* Added new event "JabberStatus" in the Jabber module to monitor buddies
@@ -1407,7 +1409,7 @@
of unit tests with the purpose of verifying the operation of C functions.
* SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
XMPP text messages to the remote JID.
- * Modules.conf has a new option - "require" - that marks a module as critical for
+ * Modules.conf has a new option - "require" - that marks a module as critical for
the execution of Asterisk.
If one of the required modules fail to load, Asterisk will exit with a return
code set to 2.
@@ -1454,7 +1456,7 @@
which applies the setting to the entire module specified, regardless of which source
files it was built from.
* New 'manager show settings' command showing the current settings loaded from
- manager.conf.
+ manager.conf.
* Added 'all' keyword to the CLI command "channel request hangup" so that you can send
the channel hangup request to all channels.
* Added a "core reload" CLI command that executes a global reload of Asterisk.
@@ -1474,7 +1476,7 @@
remote services. For backwards compatibility, "secret" still has the
same function as before, but now you can configure both a remote secret and a
local secret for mutual authentication.
- * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
+ * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
the sound will be played to the target of an attended transfer
* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
finer control over how many peers Asterisk will qualify and the gap between them
@@ -1494,7 +1496,7 @@
as a mailbox. Please see the sip.conf.sample file for more information.
* Added a function to remove SIP headers added in the dialplan before the
first INVITE is generated - SIPRemoveHeader()
- * Channel variables set with setvar= in a device configuration is now
+ * Channel variables set with setvar= in a device configuration is now
set both for inbound and outbound calls.
* Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
@@ -1650,7 +1652,7 @@
- Gives more configuration Flags for SIP-Users available (tested)
- Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
without extensibleObject (which really should be the last resort); gives
- also additional possibilities for LDAP-filter
+ also additional possibilities for LDAP-filter
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
@@ -1673,8 +1675,8 @@
* Added a new dialplan function, AST_CONFIG(), which allows you to access
variables from an Asterisk configuration file.
* The JACK_HOOK function now has a c() option to supply a custom client name.
- * Added two new dialplan functions from libspeex for audio gain control and
- denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
+ * Added two new dialplan functions from libspeex for audio gain control and
+ denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
rx directions of a channel from the dialplan.
* The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
based on other parameters. The default is still to search based on the
@@ -1733,7 +1735,7 @@
participant on the bridged channel as well.
* Chanspy has a new option, 'n', which will allow for the spied-on party's name
to be spoken instead of the channel name or number. For more information on the
- use of this option, issue the command "core show application ChanSpy" from the
+ use of this option, issue the command "core show application ChanSpy" from the
Asterisk CLI.
* Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
@@ -1743,11 +1745,11 @@
* ExternalIVR now takes several options that affect the way it performs, as
well as having several new commands. Please see the External IVR page on the Asterisk
wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
- * Added ability to communicate over a TCP socket instead of forking a child process for the
+ * Added ability to communicate over a TCP socket instead of forking a child process for the
ExternalIVR application.
* ChanIsAvail has a new option, 'a', which will return all available channels instead
of just the first one if you give the function more then one channel to check.
- * PrivacyManager now takes an option where you can specify a context where the
+ * PrivacyManager now takes an option where you can specify a context where the
given number will be matched. This way you have more control over who is allowed
and it stops the people who blindly enter 10 digits.
* ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
@@ -1770,10 +1772,10 @@
SIP Changes
-----------
* Added DNS manager support to registrations for peers referencing peer entries.
- DNS manager runs in the background which allows DNS lookups to be run asynchronously
+ DNS manager runs in the background which allows DNS lookups to be run asynchronously
as well as periodically updating the IP address. These properties allow for
better performance as well as recovery in the event of an IP change.
- * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
+ * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
load/reload of large numbers of peers/users by ~40x (for large lists of peers).
These changes also provide performance improvements for call setup and tear down.
* Added ability to specify registration expiry time on a per registration basis in
@@ -1783,8 +1785,8 @@
* Added t38pt_usertpsource option. See sip.conf.sample for details.
* Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
* 'sip show peers' and 'sip show users' display their entries sorted in
- alphabetical order, as opposed to the order they were in, in the config
- file or database.
+ alphabetical order, as opposed to the order they were in, in the config
+ file or database.
* Videosupport now supports an additional option, "always", which always sets
up video RTP ports, even on clients that don't support it. This helps with
callfiles and certain transfers to ensure that if two video phones are
@@ -1868,11 +1870,11 @@
operator. This is most helpful when working with long SQL queries in
func_odbc.conf, as the queries no longer need to be specified on a single
line.
- * CDR config file, cdr.conf, has an added option, "initiatedseconds",
+ * CDR config file, cdr.conf, has an added option, "initiatedseconds",
which will add a second to the billsec when the ending
- time is set, if the number in the microseconds field of the end time is
+ time is set, if the number in the microseconds field of the end time is
greater than the number of microseconds in the answer time. This allows
- users to count the 'initiated' seconds in their billing records.
+ users to count the 'initiated' seconds in their billing records.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
@@ -1884,7 +1886,7 @@
on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
* Manager version has changed to 1.1
* Added a new action 'CoreShowChannels' to list currently defined channels
- and some information about them.
+ and some information about them.
* Added a new action 'SIPshowregistry' to list SIP registrations.
* Added TLS support for the manager interface and HTTP server
* Added the URI redirect option for the built-in HTTP server
@@ -1896,7 +1898,7 @@
Asterisk configuration file in JSON format. This is intended to help
improve the performance of AJAX applications using the manager interface
over HTTP.
- * SIP and IAX manager events now use "ChannelType" in all cases where we
+ * SIP and IAX manager events now use "ChannelType" in all cases where we
indicate channel driver. Previously, we used a mixture of "Channel"
and "ChannelDriver" headers.
* Added a "Bridge" action which allows you to bridge any two channels that
@@ -1927,7 +1929,7 @@
* Originate now requires the Originate privilege and, if you want to call out
to a subshell, it requires the System privilege, as well. This was done to
enhance manager security.
- * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
+ * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
* New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
or manager show command Atxfer from the CLI
* New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
@@ -1949,7 +1951,7 @@
held for any given channel. Also, locks are automatically freed when a
channel is hung up.
* Added HINT() dialplan function that allows retrieving hint information.
- Hints are mappings between extensions and devices for the sake of
+ Hints are mappings between extensions and devices for the sake of
determining the state of an extension. This function can retrieve the list
of devices or the name associated with a hint.
* Added EXTENSION_STATE() dialplan function which allows retrieving the state
@@ -1999,7 +2001,7 @@
-----------
* Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
option is enabled, Asterisk will watch for a CNG tone in the incoming audio
- for a received call. If it is detected, the channel will jump to the
+ for a received call. If it is detected, the channel will jump to the
'fax' extension in the dialplan.
* The default SIP useragent= identifier now includes the Asterisk version
* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
@@ -2020,8 +2022,8 @@
* The SIPPEER function have new options for port address, call and pickup groups
* Added support for T.140 realtime text in SIP/RTP
* The "checkmwi" option has been removed from sip.conf, as it is no longer
- required due to the restructuring of how MWI is handled. See the descriptions
- in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
+ required due to the restructuring of how MWI is handled. See the descriptions
+ in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
for more information.
* Added rtpdest option to CHANNEL() dialplan function.
* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
@@ -2029,11 +2031,11 @@
in the same dial command, or if the new c option in dial() is used.
* The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
states it is not needed. For phones, however, that do require it the "registertrying" option
- has been added so it can be enabled.
+ has been added so it can be enabled.
* A new option called "callcounter" (global/peer/user level) enables call counters needed
for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
used to enable this functionality).
- * New settings for timer T1 and timer B on a global level or per device. This makes it
+ * New settings for timer T1 and timer B on a global level or per device. This makes it
possible to force timeout faster on non-responsive SIP servers. These settings are
considered advanced, so don't use them unless you have a problem.
* Added a dial string option to be able to set the To: header in an INVITE to any
@@ -2046,7 +2048,7 @@
* Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
and configs/sip.conf.sample for more information on how it is used.
* Added a new configuration option "authfailureevents" that enables manager events when
- a peer can't authenticate properly.
+ a peer can't authenticate properly.
* Added DNS manager support to registrations for peers not referencing a peer entry.
IAX2 changes
@@ -2143,7 +2145,7 @@
work with Mac CoreAudio, but portaudio supports a number of other audio
interfaces, as well. Note that this channel driver requires v19 or higher
of portaudio; older versions have a different API.
-
+
DUNDi changes
-------------
* Added the ability to specify arguments to the Dial application when using
@@ -2210,15 +2212,15 @@
-------------
* Added the general option 'shared_lastcall' so that member's wrapuptime may be
used across multiple queues.
- * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
+ * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
* Added keepstats option to queues.conf which will keep queue
statistics during a reload.
* setinterfacevar option in queues.conf also now sets a variable
called MEMBERNAME which contains the member's name.
* Added 'Strategy' field to manager event QueueParams which represents
- the queue strategy in use.
- * Added option to run macro when a queue member is connected to a caller,
+ the queue strategy in use.
+ * Added option to run macro when a queue member is connected to a caller,
see queues.conf.sample for details.
* app_queue now has a 'loose' option which is almost exactly like 'strict' except it
does not count paused queue members as unavailable.
@@ -2264,7 +2266,7 @@
MeetMe Changes
--------------
- * The 'o' option to provide an optimization has been removed and its functionality
+ * The 'o' option to provide an optimization has been removed and its functionality
has been enabled by default.
* When a conference is created, the UNIQUEID of the channel that caused it to be
created is stored. Then, every channel that joins the conference will have the
@@ -2292,7 +2294,7 @@
conference when there is only one member and the M option is used.
* Added MEETME_INFO dialplan function which provides a way to query
various properties of a Meetme conference.
- * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
+ * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
and *84: record in-conf
Other Dialplan Application Changes
@@ -2331,7 +2333,7 @@
Music On Hold Changes
---------------------
- * A new option, "digit", has been added for music on hold classes in
+ * A new option, "digit", has been added for music on hold classes in
musiconhold.conf. If this is set for a music on hold class, a caller
listening to music on hold can press this digit to switch to listening
to this music on hold class.
@@ -2344,7 +2346,7 @@
-----------
* AEL upgraded to use the Gosub with Arguments instead
of Macro application, to hopefully reduce the problems
- seen with the artificially low stack ceiling that
+ seen with the artificially low stack ceiling that
Macro bumps into. Macros can only call other Macros
to a depth of 7. Tests run using gosub, show depths
limited only by virtual memory. A small test demonstrated
@@ -2360,17 +2362,17 @@
fashion: Set(LOCAL(myvar)=someval); ("local" is now
an AEL keyword).
* utils/conf2ael introduced. Will convert an extensions.conf
- file into extensions.ael. Very crude and unfinished, but
+ file into extensions.ael. Very crude and unfinished, but
will be improved as time goes by. Should be useful for a
first pass at conversion.
* aelparse will now read extensions.conf to see if a referenced
macro or context is there before issueing a warning.
- * AEL parser sets a local channel variable ~~EXTEN~~, to
+ * AEL parser sets a local channel variable ~~EXTEN~~, to
preserve the value of ${EXTEN} thru switch statements.
* New operator in $[...] expressions: the ~~ operator serves
as a concatenation operator. AT THE MOMENT, it is really only
necessary and useful in AEL, especially in if() expressions.
- Operation: ${a} ~~ ${b| with force both a and b to strings, strip
+ Operation: ${a} ~~ ${b| with force both a and b to strings, strip
any enclosing double-quotes, and evaluate to the value of a
concatenated with the value of b. For example if a is set to
"xyz" and b has the value "abc", then ${a} ~~ ${b| would
@@ -2438,7 +2440,7 @@
and to ensure that the oldest log file gets deleted.
* Added realtime support for the queue log
-Call Detail Records
+Call Detail Records
-------------------
* The cdr_manager module has a [mappings] feature, like cdr_custom,
to add fields to the manager event from the CDR variables.
@@ -2499,7 +2501,7 @@
* Added support for writing and running your dialplan in lua using the pbx_lua
module. See configs/extensions.lua.sample for examples of how to do this.
-Miscellaneous
+Miscellaneous
-------------
* Ability to use libcap to set high ToS bits when non-root
on Linux. If configure is unable to find libcap then you
@@ -2547,7 +2549,7 @@
turned on, via the CHANNEL(trace) dialplan function. Could be useful for
dialplan debugging.
* iLBC source code no longer included (see UPGRADE.txt for details)
- * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
+ * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
deadlock is detected, a backtrace of the stack which led to the lock calls
will be output to the CLI.
* If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
Modified: trunk/build_tools/menuselect-deps.in
URL: http://svnview.digium.com/svn/asterisk/trunk/build_tools/menuselect-deps.in?view=diff&rev=370407&r1=370406&r2=370407
==============================================================================
--- trunk/build_tools/menuselect-deps.in (original)
+++ trunk/build_tools/menuselect-deps.in Mon Jul 23 16:27:56 2012
@@ -13,6 +13,7 @@
GMIME=@PBX_GMIME@
GNU_LD=@GNU_LD@
GSM=@PBX_GSM@
+ILBC=@PBX_ILBC@
GTK2=@PBX_GTK2@
H323=@PBX_H323@
HOARD=@PBX_HOARD@
Modified: trunk/codecs/Makefile
URL: http://svnview.digium.com/svn/asterisk/trunk/codecs/Makefile?view=diff&rev=370407&r1=370406&r2=370407
==============================================================================
--- trunk/codecs/Makefile (original)
+++ trunk/codecs/Makefile Mon Jul 23 16:27:56 2012
@@ -29,6 +29,12 @@
$(if $(filter codec_gsm,$(EMBEDDED_MODS)),modules.link,codec_gsm.so): gsm/lib/libgsm.a
endif
+ifneq ($(ILBC_INTERNAL),no)
+$(if $(filter codec_ilbc,$(EMBEDDED_MODS)),modules.link,codec_ilbc.so): $(LIBILBC)
+else
+ILBC_INCLUDE+=-DILBC_WEBRTC
+endif
+
clean::
$(MAKE) -C gsm clean
@@ -50,8 +56,6 @@
@$(MAKE) -C ilbc all _ASTCFLAGS="$(filter-out -Wmissing-prototypes -Wmissing-declarations -Wshadow,$(_ASTCFLAGS)) $(AST_NO_STRICT_OVERFLOW)"
-$(if $(filter codec_ilbc,$(EMBEDDED_MODS)),modules.link,codec_ilbc.so): $(LIBILBC)
-
$(if $(filter codec_g722,$(EMBEDDED_MODS)),modules.link,codec_g722.so): g722/g722_encode.o g722/g722_decode.o
g722/g722_encode.o g722/g722_decode.o: _ASTCFLAGS+=$(call MOD_ASTCFLAGS,codec_g722)
Modified: trunk/codecs/codec_ilbc.c
URL: http://svnview.digium.com/svn/asterisk/trunk/codecs/codec_ilbc.c?view=diff&rev=370407&r1=370406&r2=370407
==============================================================================
--- trunk/codecs/codec_ilbc.c (original)
+++ trunk/codecs/codec_ilbc.c Mon Jul 23 16:27:56 2012
@@ -21,11 +21,12 @@
/*! \file
*
* \brief Translate between signed linear and Internet Low Bitrate Codec
- *
+ *
* \ingroup codecs
*/
/*** MODULEINFO
+ <use>ilbc</use>
<support_level>core</support_level>
***/
@@ -37,8 +38,18 @@
#include "asterisk/module.h"
#include "asterisk/utils.h"
+#ifdef ILBC_WEBRTC
+#include <ilbc.h>
+typedef WebRtc_UWord16 ilbc_bytes;
+typedef WebRtc_Word16 ilbc_block;
+#define BUF_TYPE i16
+#else
#include "ilbc/iLBC_encode.h"
#include "ilbc/iLBC_decode.h"
+typedef unsigned char ilbc_bytes;
+typedef float ilbc_block;
+#define BUF_TYPE uc
+#endif
#define USE_ILBC_ENHANCER 0
#define ILBC_MS 30
@@ -86,7 +97,7 @@
the tail location. Read in as many frames as there are */
int x,i;
int16_t *dst = pvt->outbuf.i16;
- float tmpf[ILBC_SAMPLES];
+ ilbc_block tmpf[ILBC_SAMPLES];
if (!f->data.ptr && f->datalen) {
ast_debug(1, "issue 16070, ILIB ERROR. data = NULL datalen = %d src = %s\n", f->datalen, f->src ? f->src : "no src set");
@@ -144,13 +155,13 @@
if (pvt->samples < ILBC_SAMPLES)
return NULL;
while (pvt->samples >= ILBC_SAMPLES) {
- float tmpf[ILBC_SAMPLES];
+ ilbc_block tmpf[ILBC_SAMPLES];
int i;
/* Encode a frame of data */
for (i = 0 ; i < ILBC_SAMPLES ; i++)
tmpf[i] = tmp->buf[samples + i];
- iLBC_encode( pvt->outbuf.uc + datalen, tmpf, &tmp->enc);
+ iLBC_encode( (ilbc_bytes*)pvt->outbuf.BUF_TYPE + datalen, tmpf, &tmp->enc);
datalen += ILBC_FRAME_LEN;
samples += ILBC_SAMPLES;
Modified: trunk/configure.ac
URL: http://svnview.digium.com/svn/asterisk/trunk/configure.ac?view=diff&rev=370407&r1=370406&r2=370407
==============================================================================
--- trunk/configure.ac (original)
+++ trunk/configure.ac Mon Jul 23 16:27:56 2012
@@ -390,6 +390,7 @@
AST_EXT_LIB_SETUP([DAHDI], [DAHDI], [dahdi])
AST_EXT_LIB_SETUP([FFMPEG], [Ffmpeg and avcodec], [avcodec])
AST_EXT_LIB_SETUP([GSM], [External GSM], [gsm], [, use 'internal' GSM otherwise])
+AST_EXT_LIB_SETUP([ILBC], [System iLBC], [ilbc], [, use 'internal' iLBC otherwise])
AST_EXT_LIB_SETUP([GTK2], [gtk2], [gtk2])
AST_EXT_LIB_SETUP([GMIME], [GMime], [gmime])
AST_EXT_LIB_SETUP([OPENH323], [OpenH323], [h323])
@@ -1250,6 +1251,26 @@
fi
fi
+ILBC_INTERNAL="yes"
+AC_SUBST(ILBC_INTERNAL)
+ILBC_SYSTEM="yes"
+if test "${USE_ILBC}" != "no"; then
+ if test "${ILBC_DIR}" = "internal"; then
+ ILBC_SYSTEM="no"
+ elif test "${ILBC_DIR}" != ""; then
+ ILBC_INTERNAL="no"
+ fi
+ if test "${ILBC_SYSTEM}" = "yes"; then
+ AST_PKG_CONFIG_CHECK(ILBC, libilbc)
+ if test "$PBX_ILBC" = '1'; then
+ ILBC_INTERNAL='no'
+ fi
+ fi
+ if test "${ILBC_INTERNAL}" = "yes"; then
+ PBX_ILBC=1
+ fi
+fi
+
AST_EXT_LIB_CHECK([ICONV], [iconv], [iconv_open], [iconv.h])
# GNU libiconv #define's iconv_open to libiconv_open, so we need to search for that symbol
AST_EXT_LIB_CHECK([ICONV], [iconv], [libiconv_open], [iconv.h])
Modified: trunk/include/asterisk/autoconfig.h.in
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/autoconfig.h.in?view=diff&rev=370407&r1=370406&r2=370407
==============================================================================
--- trunk/include/asterisk/autoconfig.h.in (original)
+++ trunk/include/asterisk/autoconfig.h.in Mon Jul 23 16:27:56 2012
@@ -326,6 +326,9 @@
/* Define to 1 if you have the Iksemel Jabber library. */
#undef HAVE_IKSEMEL
+
+/* Define if your system has the ILBC libraries. */
+#undef HAVE_ILBC
/* Define if your system has the UW IMAP Toolkit c-client library. */
#undef HAVE_IMAP_TK
Modified: trunk/makeopts.in
URL: http://svnview.digium.com/svn/asterisk/trunk/makeopts.in?view=diff&rev=370407&r1=370406&r2=370407
==============================================================================
--- trunk/makeopts.in (original)
+++ trunk/makeopts.in Mon Jul 23 16:27:56 2012
@@ -141,6 +141,10 @@
GSM_INCLUDE=@GSM_INCLUDE@
GSM_LIB=@GSM_LIB@
+ILBC_INTERNAL=@ILBC_INTERNAL@
+ILBC_INCLUDE=@ILBC_INCLUDE@
+ILBC_LIB=@ILBC_LIB@
+
GTK2_INCLUDE=@GTK2_INCLUDE@
GTK2_LIB=@GTK2_LIB@
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