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https://origsvn.digium.com/svn/asterisk/trunk
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r140566 | russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines
Update instructions for getting libresample
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@140567 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines
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.
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132391 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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r132388 | russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines
Enable higher quality resampling, as it doesn't have a noticeable performance
impact on my machine ..
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132389 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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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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