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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 43a78100c0
commit 28501051b4
5 changed files with 134 additions and 31 deletions
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63
main/codec_builtin.c
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@@ -772,6 +772,65 @@ static struct ast_codec t140 = {
.type = AST_MEDIA_TYPE_TEXT,
};
static int silk_samples(struct ast_frame *frame)
{
/* XXX This is likely not at all what's intended from this callback. However,
* since SILK is variable bit rate, I have no idea how to take a frame of data
* and determine the number of samples present. Instead, we base this on the
* sample rate of the codec and the expected number of samples to receive in 20ms.
* In testing, this has worked just fine.
*/
return ast_format_get_sample_rate(frame->subclass.format) / 50;
}
static struct ast_codec silk8 = {
.name = "silk",
.description = "SILK Codec (8 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 160,
.samples_count = silk_samples
};
static struct ast_codec silk12 = {
.name = "silk",
.description = "SILK Codec (12 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 12000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 240,
.samples_count = silk_samples
};
static struct ast_codec silk16 = {
.name = "silk",
.description = "SILK Codec (16 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 16000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 320,
.samples_count = silk_samples
};
static struct ast_codec silk24 = {
.name = "silk",
.description = "SILK Codec (24 KHz)",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 24000,
.minimum_ms = 20,
.maximum_ms = 100,
.default_ms = 20,
.minimum_bytes = 480,
.samples_count = silk_samples
};
#define CODEC_REGISTER_AND_CACHE(codec) \
({ \
int __res_ ## __LINE__ = 0; \
@@ -843,6 +902,10 @@ int ast_codec_builtin_init(void)
res |= CODEC_REGISTER_AND_CACHE(t140red);
res |= CODEC_REGISTER_AND_CACHE(t140);
res |= CODEC_REGISTER_AND_CACHE(none);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk8", silk8);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk12", silk12);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk16", silk16);
res |= CODEC_REGISTER_AND_CACHE_NAMED("silk24", silk24);
return res;
}