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

..

asterisk

Update support for SILK format.

2016-07-14 15:59:49 -05:00

solaris-compat

fix the provided unsetenv for solaris to return an int like it's supposed to

2006-03-29 04:14:12 +00:00

asterisk.h

Modules: Make ast_module_info->self available to auxiliary sources.

2015-05-04 20:47:01 -04:00

jitterbuf.h

incorporates r159808 from branches/1.4:

2008-11-29 17:57:39 +00:00