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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
parent
b6a0ae0b35
commit
b5193428a7
112
CHANGES
112
CHANGES
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@ -78,7 +78,7 @@ ExternalIVR
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-------------------
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* Added support for IPv6.
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* Add interrupt ('I') command to ExternalIVR. Sending this command from an
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* Add interrupt ('I') command to ExternalIVR. Sending this command from an
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external process will cause the current playlist to be cleared, including
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stopping any audio file that is currently playing. This is useful when you
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want to interrupt audio playback only when specific DTMF is entered by the
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@ -254,7 +254,7 @@ chan_ooh323
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* Added NAT support for RTP. Setting in config is 'nat', which can be set
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globally and overriden on a peer by peer basis.
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* Direct media functionality has been added. Options in config are:
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* Direct media functionality has been added. Options in config are:
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directmedia (directrtp) and directrtpsetup (earlydirect)
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* ChannelUpdate events now contain a CallRef header.
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@ -354,16 +354,16 @@ chan_unistim
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* Added global 'debug' option, that enables debug in channel driver
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* Added ability to translate on-screen menu in multiple languages. Tested on
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Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
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ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
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Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
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ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
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menu of phone
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* In addition to English added French and Russian languages for on-screen menus
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* Reworked dialing number input: added dialing by timeout, immediate dial on
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* Reworked dialing number input: added dialing by timeout, immediate dial on
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on dialplan compare, phone number length now not limited by screen size
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* Added ability to pickup a call using features.conf defined value and
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* Added ability to pickup a call using features.conf defined value and
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on-screen key
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@ -421,7 +421,7 @@ AMI (Asterisk Manager Interface)
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returned if auto_force_rport is not enabled.
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* Hangup now can take a regular expression as the Channel option. If you want
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to hangup multiple channels, use /regex/ as the Channel option. Existing
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to hangup multiple channels, use /regex/ as the Channel option. Existing
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behavior to hanging up a single channel is unchanged, but if you pass a regex,
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the manager will send you a list of channels back that were hung up.
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@ -509,6 +509,8 @@ Codecs
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and CELT. You are able to set up a call and have attribute information pass.
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This should help considerably with video calls.
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* The iLBC codec can now use a system-provided iLBC library if one is installed,
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just like the GSM codec.
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Logging
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-------------------
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@ -545,7 +547,7 @@ Parking
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* Channel variable PARKER is now set when comebacktoorigin is disabled in
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a parking lot.
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* Channel variable PARKEDCALL is now set with the name of the parking lot
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* Channel variable PARKEDCALL is now set with the name of the parking lot
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when a timeout occurs.
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@ -567,7 +569,7 @@ Resource Modules
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Calendars
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-------------------
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* A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
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* A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
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CALENDAR_WRITE has completed successfully.
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@ -711,7 +713,7 @@ ConfBridge
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mixing audio at sample rates ranging from 8khz-96khz.
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* CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
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and bridge profiles on a channel.
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* CONFBRIDGE_INFO dialplan function capable of retrieving information
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* CONFBRIDGE_INFO dialplan function capable of retrieving information
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about a conference such as locked status and number of parties, admins,
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and marked users.
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* Addition of video_mode option in confbridge.conf for adding video support
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@ -759,7 +761,7 @@ Calendaring
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MixMonitor
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--------------------------
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* Added two new options, r and t with file name arguments to record
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* Added two new options, r and t with file name arguments to record
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single direction (unmixed) audio recording separate from the bidirectional
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(mixed) recording. The mixed file name argument is optional now as long
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as at least one recording option is used.
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@ -1023,7 +1025,7 @@ Applications
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* Added 'y' option to app_record. This option enables a mode where any DTMF digit
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received will terminate recording.
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* Voicemail now supports per mailbox settings for folders when using IMAP storage.
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Previously the folder could only be set per context, but has now been extended
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Previously the folder could only be set per context, but has now been extended
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using the imapfolder option.
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* Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
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* Voicemail now allows the pager date format to be specified separately from the
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@ -1274,7 +1276,7 @@ libpri channel driver (chan_dahdi) DAHDI changes
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to eliminate tromboned calls. A tromboned call goes out an interface and comes
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back into the same interface. Tromboned calls happen because of call routing,
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call deflection, call forwarding, and call transfer.
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* Added the ability to send and receive ETSI Advice-Of-Charge messages.
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* Added the ability to send and receive ETSI Advice-Of-Charge messages.
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* Added the ability to support call waiting calls. (The SETUP has no B channel
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assigned.)
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* Added Malicious Call ID (MCID) event to the AMI call event class.
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@ -1298,7 +1300,7 @@ Asterisk Manager Interface
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* The configuration file manager.conf now supports a channelvars option, which
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specifies a list of channel variables to include in each channel-oriented
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event.
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* The redirect command now has new parameters ExtraContext, ExtraExtension,
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* The redirect command now has new parameters ExtraContext, ExtraExtension,
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and ExtraPriority to allow redirecting the second channel to a different
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location than the first.
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* Added new event "JabberStatus" in the Jabber module to monitor buddies
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@ -1407,7 +1409,7 @@ Miscellaneous
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of unit tests with the purpose of verifying the operation of C functions.
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* SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
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XMPP text messages to the remote JID.
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* Modules.conf has a new option - "require" - that marks a module as critical for
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* Modules.conf has a new option - "require" - that marks a module as critical for
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the execution of Asterisk.
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If one of the required modules fail to load, Asterisk will exit with a return
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code set to 2.
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@ -1454,7 +1456,7 @@ CLI Changes
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which applies the setting to the entire module specified, regardless of which source
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files it was built from.
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* New 'manager show settings' command showing the current settings loaded from
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manager.conf.
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manager.conf.
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* Added 'all' keyword to the CLI command "channel request hangup" so that you can send
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the channel hangup request to all channels.
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* Added a "core reload" CLI command that executes a global reload of Asterisk.
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@ -1474,7 +1476,7 @@ SIP Changes
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remote services. For backwards compatibility, "secret" still has the
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same function as before, but now you can configure both a remote secret and a
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local secret for mutual authentication.
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* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
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* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
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the sound will be played to the target of an attended transfer
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* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
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finer control over how many peers Asterisk will qualify and the gap between them
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@ -1494,7 +1496,7 @@ SIP Changes
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as a mailbox. Please see the sip.conf.sample file for more information.
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* Added a function to remove SIP headers added in the dialplan before the
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first INVITE is generated - SIPRemoveHeader()
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* Channel variables set with setvar= in a device configuration is now
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* Channel variables set with setvar= in a device configuration is now
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set both for inbound and outbound calls.
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* Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
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@ -1650,7 +1652,7 @@ LDAP Schema File Additions
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- Gives more configuration Flags for SIP-Users available (tested)
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- Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
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without extensibleObject (which really should be the last resort); gives
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also additional possibilities for LDAP-filter
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also additional possibilities for LDAP-filter
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
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@ -1673,8 +1675,8 @@ Dialplan Functions
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* Added a new dialplan function, AST_CONFIG(), which allows you to access
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variables from an Asterisk configuration file.
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* The JACK_HOOK function now has a c() option to supply a custom client name.
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* Added two new dialplan functions from libspeex for audio gain control and
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denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
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* Added two new dialplan functions from libspeex for audio gain control and
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denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
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rx directions of a channel from the dialplan.
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* The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
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based on other parameters. The default is still to search based on the
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@ -1733,7 +1735,7 @@ Application Changes
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participant on the bridged channel as well.
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* Chanspy has a new option, 'n', which will allow for the spied-on party's name
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to be spoken instead of the channel name or number. For more information on the
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use of this option, issue the command "core show application ChanSpy" from the
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use of this option, issue the command "core show application ChanSpy" from the
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Asterisk CLI.
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* Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
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spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
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@ -1743,11 +1745,11 @@ Application Changes
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* ExternalIVR now takes several options that affect the way it performs, as
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well as having several new commands. Please see the External IVR page on the Asterisk
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wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
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* Added ability to communicate over a TCP socket instead of forking a child process for the
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* Added ability to communicate over a TCP socket instead of forking a child process for the
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ExternalIVR application.
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* ChanIsAvail has a new option, 'a', which will return all available channels instead
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of just the first one if you give the function more then one channel to check.
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* PrivacyManager now takes an option where you can specify a context where the
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* PrivacyManager now takes an option where you can specify a context where the
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given number will be matched. This way you have more control over who is allowed
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and it stops the people who blindly enter 10 digits.
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* ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
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@ -1770,10 +1772,10 @@ Application Changes
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SIP Changes
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-----------
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* Added DNS manager support to registrations for peers referencing peer entries.
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DNS manager runs in the background which allows DNS lookups to be run asynchronously
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DNS manager runs in the background which allows DNS lookups to be run asynchronously
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as well as periodically updating the IP address. These properties allow for
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better performance as well as recovery in the event of an IP change.
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* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
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* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
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load/reload of large numbers of peers/users by ~40x (for large lists of peers).
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These changes also provide performance improvements for call setup and tear down.
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* Added ability to specify registration expiry time on a per registration basis in
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@ -1783,8 +1785,8 @@ SIP Changes
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* Added t38pt_usertpsource option. See sip.conf.sample for details.
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* Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
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* 'sip show peers' and 'sip show users' display their entries sorted in
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alphabetical order, as opposed to the order they were in, in the config
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file or database.
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alphabetical order, as opposed to the order they were in, in the config
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file or database.
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* Videosupport now supports an additional option, "always", which always sets
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up video RTP ports, even on clients that don't support it. This helps with
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callfiles and certain transfers to ensure that if two video phones are
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@ -1868,11 +1870,11 @@ Miscellaneous
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operator. This is most helpful when working with long SQL queries in
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func_odbc.conf, as the queries no longer need to be specified on a single
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line.
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* CDR config file, cdr.conf, has an added option, "initiatedseconds",
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* CDR config file, cdr.conf, has an added option, "initiatedseconds",
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which will add a second to the billsec when the ending
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time is set, if the number in the microseconds field of the end time is
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time is set, if the number in the microseconds field of the end time is
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greater than the number of microseconds in the answer time. This allows
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users to count the 'initiated' seconds in their billing records.
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users to count the 'initiated' seconds in their billing records.
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
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@ -1884,7 +1886,7 @@ AMI - The manager (TCP/TLS/HTTP)
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on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
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* Manager version has changed to 1.1
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* Added a new action 'CoreShowChannels' to list currently defined channels
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and some information about them.
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and some information about them.
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* Added a new action 'SIPshowregistry' to list SIP registrations.
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* Added TLS support for the manager interface and HTTP server
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* Added the URI redirect option for the built-in HTTP server
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@ -1896,7 +1898,7 @@ AMI - The manager (TCP/TLS/HTTP)
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Asterisk configuration file in JSON format. This is intended to help
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improve the performance of AJAX applications using the manager interface
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over HTTP.
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* SIP and IAX manager events now use "ChannelType" in all cases where we
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* SIP and IAX manager events now use "ChannelType" in all cases where we
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indicate channel driver. Previously, we used a mixture of "Channel"
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and "ChannelDriver" headers.
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* Added a "Bridge" action which allows you to bridge any two channels that
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@ -1927,7 +1929,7 @@ AMI - The manager (TCP/TLS/HTTP)
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* Originate now requires the Originate privilege and, if you want to call out
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to a subshell, it requires the System privilege, as well. This was done to
|
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enhance manager security.
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* Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
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* Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
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||||
* New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
|
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or manager show command Atxfer from the CLI
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* New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
|
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@ -1949,7 +1951,7 @@ Dialplan functions
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held for any given channel. Also, locks are automatically freed when a
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channel is hung up.
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* Added HINT() dialplan function that allows retrieving hint information.
|
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Hints are mappings between extensions and devices for the sake of
|
||||
Hints are mappings between extensions and devices for the sake of
|
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determining the state of an extension. This function can retrieve the list
|
||||
of devices or the name associated with a hint.
|
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* Added EXTENSION_STATE() dialplan function which allows retrieving the state
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||||
|
|
@ -1999,7 +2001,7 @@ SIP changes
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-----------
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* 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
|
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for a received call. If it is detected, the channel will jump to the
|
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for a received call. If it is detected, the channel will jump to the
|
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'fax' extension in the dialplan.
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* The default SIP useragent= identifier now includes the Asterisk version
|
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* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
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@ -2020,8 +2022,8 @@ SIP changes
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* The SIPPEER function have new options for port address, call and pickup groups
|
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* Added support for T.140 realtime text in SIP/RTP
|
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* 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
|
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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
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||||
for more information.
|
||||
* Added rtpdest option to CHANNEL() dialplan function.
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||||
* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
|
||||
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|
@ -2029,11 +2031,11 @@ SIP changes
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|||
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 @@ SIP changes
|
|||
* 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 @@ New Channel Drivers
|
|||
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 @@ Queue changes
|
|||
-------------
|
||||
* 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 @@ Queue changes
|
|||
|
||||
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 @@ MeetMe Changes
|
|||
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 @@ Other Dialplan Application Changes
|
|||
|
||||
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 Changes
|
|||
-----------
|
||||
* 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 @@ AEL Changes
|
|||
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 @@ Logger changes
|
|||
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 @@ Miscellaneous New Modules
|
|||
* 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 @@ Miscellaneous
|
|||
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
|
||||
|
|
|
|||
|
|
@ -13,6 +13,7 @@ GENERIC_ODBC=@PBX_GENERIC_ODBC@
|
|||
GMIME=@PBX_GMIME@
|
||||
GNU_LD=@GNU_LD@
|
||||
GSM=@PBX_GSM@
|
||||
ILBC=@PBX_ILBC@
|
||||
GTK2=@PBX_GTK2@
|
||||
H323=@PBX_H323@
|
||||
HOARD=@PBX_HOARD@
|
||||
|
|
|
|||
|
|
@ -29,6 +29,12 @@ GSM_INCLUDE:=-Igsm/inc
|
|||
$(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 @@ $(LIBILBC):
|
|||
@$(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+=$(call MOD_ASTCFLAGS,codec_g722)
|
||||
|
||||
|
|
|
|||
|
|
@ -21,11 +21,12 @@
|
|||
/*! \file
|
||||
*
|
||||
* \brief Translate between signed linear and Internet Low Bitrate Codec
|
||||
*
|
||||
*
|
||||
* \ingroup codecs
|
||||
*/
|
||||
|
||||
/*** MODULEINFO
|
||||
<use>ilbc</use>
|
||||
<support_level>core</support_level>
|
||||
***/
|
||||
|
||||
|
|
@ -37,8 +38,18 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
|||
#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 @@ static int ilbctolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
|
|||
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 @@ static struct ast_frame *lintoilbc_frameout(struct ast_trans_pvt *pvt)
|
|||
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;
|
||||
|
|
|
|||
21
configure.ac
21
configure.ac
|
|
@ -390,6 +390,7 @@ AST_EXT_LIB_SETUP([CRYPTO], [OpenSSL Cryptography], [crypto])
|
|||
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 @@ if test "${USE_GSM}" != "no"; then
|
|||
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])
|
||||
|
|
|
|||
|
|
@ -327,6 +327,9 @@
|
|||
/* 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
|
||||
|
||||
|
|
|
|||
|
|
@ -141,6 +141,10 @@ GSM_INTERNAL=@GSM_INTERNAL@
|
|||
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@
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue