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

main/codec_builtin.c

@@ -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;
 }

main/format_cache.c

@@ -232,6 +232,14 @@ struct ast_format *ast_format_t140_red;
  */
 struct ast_format *ast_format_none;
 
+/*!
+ * \brief Built-in "silk" format
+ */
+struct ast_format *ast_format_silk8;
+struct ast_format *ast_format_silk12;
+struct ast_format *ast_format_silk16;
+struct ast_format *ast_format_silk24;
+
 /*! \brief Number of buckets to use for the media format cache (should be prime for performance reasons) */
 #define CACHE_BUCKETS 53
 
@@ -331,6 +339,10 @@ static void format_cache_shutdown(void)
 	ao2_replace(ast_format_t140_red, NULL);
 	ao2_replace(ast_format_t140, NULL);
 	ao2_replace(ast_format_none, NULL);
+	ao2_replace(ast_format_silk8, NULL);
+	ao2_replace(ast_format_silk12, NULL);
+	ao2_replace(ast_format_silk16, NULL);
+	ao2_replace(ast_format_silk24, NULL);
 }
 
 int ast_format_cache_init(void)
@@ -426,6 +438,14 @@ static void set_cached_format(const char *name, struct ast_format *format)
 		ao2_replace(ast_format_t140, format);
 	} else if (!strcmp(name, "none")) {
 		ao2_replace(ast_format_none, format);
+	} else if (!strcmp(name, "silk8")) {
+		ao2_replace(ast_format_silk8, format);
+	} else if (!strcmp(name, "silk12")) {
+		ao2_replace(ast_format_silk12, format);
+	} else if (!strcmp(name, "silk16")) {
+		ao2_replace(ast_format_silk16, format);
+	} else if (!strcmp(name, "silk24")) {
+		ao2_replace(ast_format_silk24, format);
 	}
 }
 

main/rtp_engine.c

@@ -2198,6 +2198,11 @@ int ast_rtp_engine_init(void)
 	/* Opus and VP8 */
 	set_next_mime_type(ast_format_opus, 0,  "audio", "opus", 48000);
 	set_next_mime_type(ast_format_vp8, 0,  "video", "VP8", 90000);
+	/* DA SILK */
+	set_next_mime_type(ast_format_silk8, 0, "audio", "silk", 8000);
+	set_next_mime_type(ast_format_silk12, 0, "audio", "silk", 12000);
+	set_next_mime_type(ast_format_silk16, 0, "audio", "silk", 16000);
+	set_next_mime_type(ast_format_silk24, 0, "audio", "silk", 24000);
 
 	/* Define the static rtp payload mappings */
 	add_static_payload(0, ast_format_ulaw, 0);
@@ -2243,6 +2248,11 @@ int ast_rtp_engine_init(void)
 	add_static_payload(100, ast_format_vp8, 0);
 	add_static_payload(107, ast_format_opus, 0);
 
+	add_static_payload(108, ast_format_silk8, 0);
+	add_static_payload(109, ast_format_silk12, 0);
+	add_static_payload(113, ast_format_silk16, 0);
+	add_static_payload(114, ast_format_silk24, 0);
+
 	return 0;
 }