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| author | Mark Michelson <mmichelson@digium.com> | 2016-06-30 15:58:53 -0500 |
|---|---|---|
| committer | Mark Michelson <mmichelson@digium.com> | 2016-07-14 15:59:49 -0500 |
| commit | 273052f40498378d3f2d3548347a243df68ee9a4 (patch) | |
| tree | 3d959becc0aa82c887a4b70643d5481a5ce7c797 /utils | |
| parent | 3cf33dd4e7e886383531335efda3baca728b1f51 (diff) | |
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
Diffstat (limited to 'utils')
0 files changed, 0 insertions, 0 deletions