Commit Graph

12 Commits

Author SHA1 Message Date
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
2008-12-15 14:40:08 +00:00
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
2008-11-01 21:10:07 +00:00
Russell Bryant c0c743a5fa Update instructions for getting libresample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-02 15:11:53 +00:00
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
2008-08-07 17:26:23 +00:00
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
2008-07-21 14:47:41 +00:00
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
2008-07-21 13:51:05 +00:00
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
2008-05-22 17:10:53 +00:00
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
2008-05-13 17:42:17 +00:00
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
2008-05-10 03:28:50 +00:00
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
2008-04-22 16:47:00 +00:00
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
2008-01-14 04:53:08 +00:00
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
2008-01-13 19:19:57 +00:00