Files
asterisk/res
Mark Michelson 273052f404 Update support for SILK format.
This commit adds scaffolding in order to support the SILK audio format
on calls. Roughly, this is what is added:

* Cached silk formats. One for each possible sample rate.
* ast_codec structures for each possible sample rate.
* RTP payload mappings for "SILK".

In addition, this change overhauls the res_format_attr_silk file in the
following ways:

* The "samplerate" attribute is scrapped. That's native to the format.
* There are far more checks to ensure that attributes have been
  allocated before attempting to reference them.
* We do not SDP fmtp lines for attributes set to 0.

These changes make way to be able to install a codec_silk module and
have it actually work. It also should allow for passthrough silk calls
in Asterisk.

Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
2016-07-14 15:59:49 -05:00
..
2016-03-29 09:03:55 -05:00
2015-11-24 13:57:05 -06:00
2016-06-08 20:37:08 +03:00
2016-07-12 05:00:16 -05:00
2016-03-29 09:03:55 -05:00
2015-05-19 21:11:21 -05:00