asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	e612d87695	Convert a few places to use ast_calloc_with_stringfields where applicable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	a05f90e709	minor tweak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	338e22b866	Constify a string and strip trailing whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) 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: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	bb03eef676	Making sure we have references to external libraries. Note: Update h.323 with the recent changes too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	bca058070e	Fix build WRT ast_str_opaque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b168ee34b	Merge changes from team/group/appdocsxml 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: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c0c743a5fa	Update instructions for getting libresample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ec4952cf73	stop using deprecated API call git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c87f901cfd	Remove libresample from the Asterisk source tree. It is now available in its 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: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5de127e103	Enable higher quality resampling, as it doesn't have a noticeable performance impact on my machine .. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	f7eb823a7a	Fix a few places where frame data was used directly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	ea3fb96b29	Re-introduce proper error handling that was removed in recent commits. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Claude Patry	dfe475cc59	ameliorate load and unload to dont use DECLINED or FAILED, when theres no .conf involved. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	01f3a08f8a	Add a c() option for the Jack() application and JACK_HOOK() funciton for 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: https://origsvn.digium.com/svn/asterisk/trunk@114533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	577666bca0	Add another small option for the JACK app and JACK_HOOK function. The 'n' 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: https://origsvn.digium.com/svn/asterisk/trunk@98676 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f32aec9f8f	Bring in the code from team/russell/jack/. 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: https://origsvn.digium.com/svn/asterisk/trunk@98628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

18 Commits (7d96f2d01e1b00037c0574667866937b08572856)