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Update support for SILK format.

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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
This commit is contained in:
Mark Michelson
2016-06-30 15:58:53 -05:00
parent 3cf33dd4e7
commit 273052f404
5 changed files with 134 additions and 31 deletions
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8
include/asterisk/format_cache.h
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@@ -223,6 +223,14 @@ extern struct ast_format *ast_format_t140_red;
*/
extern struct ast_format *ast_format_none;
/*!
* \brief Built-in SILK format.
*/
extern struct ast_format *ast_format_silk8;
extern struct ast_format *ast_format_silk12;
extern struct ast_format *ast_format_silk16;
extern struct ast_format *ast_format_silk24;
/*!
* \brief Initialize format cache support within the core.
*