[Asterisk-code-review] Update support for SILK format. (asterisk[13])

[Mark Michelson]

Mark Michelson has uploaded a new patch set (#2).

Change subject: Update support for SILK format.
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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
---
M include/asterisk/format_cache.h
M main/codec_builtin.c
M main/format_cache.c
M main/rtp_engine.c
M res/res_format_attr_silk.c
5 files changed, 155 insertions(+), 23 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/36/3136/2
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>