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2010-01-15Convert a few places to use ast_calloc_with_stringfields where applicable.seanbright1-5/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240368 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman1-5/+5
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-06minor tweakrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199372 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-06Constify a string and strip trailing whitespace.russell1-23/+24
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199370 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-7/+2
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-14Making sure we have references to external libraries.oej1-1/+1
Note: Update h.323 with the recent changes too git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188283 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Fix build WRT ast_str_opaquerussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164202 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell1-12/+40
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-02Update instructions for getting libresamplerussell1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140566 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07stop using deprecated API callkpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136504 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Remove libresample from the Asterisk source tree. It is now available in itsrussell1-1/+4
own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132390 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Enable higher quality resampling, as it doesn't have a noticeable performancerussell1-1/+1
impact on my machine .. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132388 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22Fix a few places where frame data was used directly.qwell1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117828 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-13Re-introduce proper error handling that was removed in recent commits.russell1-4/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115850 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-10ameliorate load and unload to dont use DECLINED or FAILED, when theres no ↵junky1-7/+4
.conf involved. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115594 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-22Add a c() option for the Jack() application and JACK_HOOK() funciton for ↵russell1-7/+27
supplying a custom client name. Using the channel name is still the default. This was done at the request of Jared Smith. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114533 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Add another small option for the JACK app and JACK_HOOK function. The 'n'russell1-6/+18
option tells JACK not to start jackd automatically if it is not already running. Otherwise, the default is that jackd will get started for you if it isn't running already. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98676 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-13Bring in the code from team/russell/jack/.russell1-0/+959
Add a new module, app_jack, which provides interfaces to JACK, the Jack Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are provided; there is a JACK() application, and a JACK_HOOK() function. Both interfaces create an input and output JACK port. The application makes these ports the endpoint of the call. The audio coming from the channel goes out the output port and whatever comes back in on the input port is what gets sent to the channel. The JACK_HOOK() function turns on a JACK audiohook on the channel. This lets you run the audio coming from a channel through JACK, and whatever comes back in is what gets forwarded on as the channel's audio. This is very useful for building custom vocoders or doing recording or analysis of the channel's audio in another application. In case anyone is curious, the platform that inspired me to write this is PureData (http://puredata.info/). I wrote these JACK interfaces so that I could use Pd to do interesting things with the audio of phone calls ... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98628 f38db490-d61c-443f-a65b-d21fe96a405b
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