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https://github.com/asterisk/asterisk.git
synced 2025-09-03 19:28:53 +00:00
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
This commit is contained in:
@@ -223,6 +223,14 @@ extern struct ast_format *ast_format_t140_red;
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*/
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extern struct ast_format *ast_format_none;
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/*!
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* \brief Built-in SILK format.
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*/
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extern struct ast_format *ast_format_silk8;
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extern struct ast_format *ast_format_silk12;
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extern struct ast_format *ast_format_silk16;
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extern struct ast_format *ast_format_silk24;
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/*!
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* \brief Initialize format cache support within the core.
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*
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@@ -772,6 +772,65 @@ static struct ast_codec t140 = {
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.type = AST_MEDIA_TYPE_TEXT,
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};
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static int silk_samples(struct ast_frame *frame)
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{
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/* XXX This is likely not at all what's intended from this callback. However,
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* since SILK is variable bit rate, I have no idea how to take a frame of data
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* and determine the number of samples present. Instead, we base this on the
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* sample rate of the codec and the expected number of samples to receive in 20ms.
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* In testing, this has worked just fine.
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*/
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return ast_format_get_sample_rate(frame->subclass.format) / 50;
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}
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static struct ast_codec silk8 = {
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.name = "silk",
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.description = "SILK Codec (8 KHz)",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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.minimum_ms = 20,
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.maximum_ms = 100,
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.default_ms = 20,
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.minimum_bytes = 160,
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.samples_count = silk_samples
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};
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static struct ast_codec silk12 = {
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.name = "silk",
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.description = "SILK Codec (12 KHz)",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 12000,
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.minimum_ms = 20,
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.maximum_ms = 100,
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.default_ms = 20,
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.minimum_bytes = 240,
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.samples_count = silk_samples
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};
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static struct ast_codec silk16 = {
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.name = "silk",
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.description = "SILK Codec (16 KHz)",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 16000,
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.minimum_ms = 20,
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.maximum_ms = 100,
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.default_ms = 20,
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.minimum_bytes = 320,
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.samples_count = silk_samples
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};
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static struct ast_codec silk24 = {
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.name = "silk",
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.description = "SILK Codec (24 KHz)",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 24000,
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.minimum_ms = 20,
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.maximum_ms = 100,
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.default_ms = 20,
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.minimum_bytes = 480,
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.samples_count = silk_samples
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};
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#define CODEC_REGISTER_AND_CACHE(codec) \
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({ \
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int __res_ ## __LINE__ = 0; \
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@@ -843,6 +902,10 @@ int ast_codec_builtin_init(void)
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res |= CODEC_REGISTER_AND_CACHE(t140red);
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res |= CODEC_REGISTER_AND_CACHE(t140);
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res |= CODEC_REGISTER_AND_CACHE(none);
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res |= CODEC_REGISTER_AND_CACHE_NAMED("silk8", silk8);
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res |= CODEC_REGISTER_AND_CACHE_NAMED("silk12", silk12);
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res |= CODEC_REGISTER_AND_CACHE_NAMED("silk16", silk16);
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res |= CODEC_REGISTER_AND_CACHE_NAMED("silk24", silk24);
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return res;
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}
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@@ -232,6 +232,14 @@ struct ast_format *ast_format_t140_red;
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*/
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struct ast_format *ast_format_none;
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/*!
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* \brief Built-in "silk" format
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*/
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struct ast_format *ast_format_silk8;
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struct ast_format *ast_format_silk12;
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struct ast_format *ast_format_silk16;
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struct ast_format *ast_format_silk24;
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/*! \brief Number of buckets to use for the media format cache (should be prime for performance reasons) */
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#define CACHE_BUCKETS 53
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@@ -331,6 +339,10 @@ static void format_cache_shutdown(void)
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ao2_replace(ast_format_t140_red, NULL);
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ao2_replace(ast_format_t140, NULL);
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ao2_replace(ast_format_none, NULL);
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ao2_replace(ast_format_silk8, NULL);
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ao2_replace(ast_format_silk12, NULL);
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ao2_replace(ast_format_silk16, NULL);
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ao2_replace(ast_format_silk24, NULL);
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}
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int ast_format_cache_init(void)
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@@ -426,6 +438,14 @@ static void set_cached_format(const char *name, struct ast_format *format)
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ao2_replace(ast_format_t140, format);
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} else if (!strcmp(name, "none")) {
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ao2_replace(ast_format_none, format);
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} else if (!strcmp(name, "silk8")) {
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ao2_replace(ast_format_silk8, format);
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} else if (!strcmp(name, "silk12")) {
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ao2_replace(ast_format_silk12, format);
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} else if (!strcmp(name, "silk16")) {
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ao2_replace(ast_format_silk16, format);
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} else if (!strcmp(name, "silk24")) {
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ao2_replace(ast_format_silk24, format);
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}
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}
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@@ -2690,6 +2690,11 @@ int ast_rtp_engine_init(void)
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/* Opus and VP8 */
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set_next_mime_type(ast_format_opus, 0, "audio", "opus", 48000);
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set_next_mime_type(ast_format_vp8, 0, "video", "VP8", 90000);
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/* DA SILK */
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set_next_mime_type(ast_format_silk8, 0, "audio", "silk", 8000);
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set_next_mime_type(ast_format_silk12, 0, "audio", "silk", 12000);
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set_next_mime_type(ast_format_silk16, 0, "audio", "silk", 16000);
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set_next_mime_type(ast_format_silk24, 0, "audio", "silk", 24000);
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/* Define the static rtp payload mappings */
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add_static_payload(0, ast_format_ulaw, 0);
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@@ -2743,6 +2748,11 @@ int ast_rtp_engine_init(void)
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add_static_payload(100, ast_format_vp8, 0);
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add_static_payload(107, ast_format_opus, 0);
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add_static_payload(108, ast_format_silk8, 0);
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add_static_payload(109, ast_format_silk12, 0);
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add_static_payload(113, ast_format_silk16, 0);
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add_static_payload(114, ast_format_silk24, 0);
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return 0;
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}
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@@ -40,7 +40,6 @@ ASTERISK_REGISTER_FILE()
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* \note The only attribute that affects compatibility here is the sample rate.
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*/
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struct silk_attr {
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unsigned int samplerate;
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unsigned int maxbitrate;
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unsigned int dtx;
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unsigned int fec;
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@@ -54,10 +53,15 @@ static void silk_destroy(struct ast_format *format)
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ast_free(attr);
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}
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static void attr_init(struct silk_attr *attr)
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{
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memset(attr, 0, sizeof(*attr));
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}
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static int silk_clone(const struct ast_format *src, struct ast_format *dst)
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{
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struct silk_attr *original = ast_format_get_attribute_data(src);
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struct silk_attr *attr = ast_calloc(1, sizeof(*attr));
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struct silk_attr *attr = ast_malloc(sizeof(*attr));
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if (!attr) {
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return -1;
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@@ -65,6 +69,8 @@ static int silk_clone(const struct ast_format *src, struct ast_format *dst)
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if (original) {
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*attr = *original;
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} else {
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attr_init(attr);
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}
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ast_format_set_attribute_data(dst, attr);
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@@ -109,17 +115,17 @@ static void silk_generate_sdp_fmtp(const struct ast_format *format, unsigned int
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ast_str_append(str, 0, "a=fmtp:%u maxaveragebitrate=%u\r\n", payload, attr->maxbitrate);
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}
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if (attr->dtx) {
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ast_str_append(str, 0, "a=fmtp:%u usedtx=%u\r\n", payload, attr->dtx);
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}
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if (attr->fec) {
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ast_str_append(str, 0, "a=fmtp:%u useinbandfec=%u\r\n", payload, attr->fec);
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}
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}
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static enum ast_format_cmp_res silk_cmp(const struct ast_format *format1, const struct ast_format *format2)
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{
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struct silk_attr *attr1 = ast_format_get_attribute_data(format1);
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struct silk_attr *attr2 = ast_format_get_attribute_data(format2);
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if (((!attr1 || !attr1->samplerate) && (!attr2 || !attr2->samplerate)) ||
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(attr1->samplerate == attr2->samplerate)) {
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if (ast_format_get_sample_rate(format1) == ast_format_get_sample_rate(format2)) {
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return AST_FORMAT_CMP_EQUAL;
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}
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@@ -130,13 +136,10 @@ static struct ast_format *silk_getjoint(const struct ast_format *format1, const
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{
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struct silk_attr *attr1 = ast_format_get_attribute_data(format1);
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struct silk_attr *attr2 = ast_format_get_attribute_data(format2);
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unsigned int samplerate;
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struct ast_format *jointformat;
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struct silk_attr *attr_res;
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samplerate = attr1->samplerate & attr2->samplerate;
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/* sample rate is the only attribute that has any bearing on if joint capabilities exist or not */
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if (samplerate) {
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if (ast_format_get_sample_rate(format1) != ast_format_get_sample_rate(format2)) {
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return NULL;
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}
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@@ -145,8 +148,10 @@ static struct ast_format *silk_getjoint(const struct ast_format *format1, const
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return NULL;
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}
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attr_res = ast_format_get_attribute_data(jointformat);
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attr_res->samplerate = samplerate;
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if (!attr1 || !attr2) {
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attr_init(attr_res);
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} else {
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/* Take the lowest max bitrate */
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attr_res->maxbitrate = MIN(attr1->maxbitrate, attr2->maxbitrate);
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@@ -161,6 +166,7 @@ static struct ast_format *silk_getjoint(const struct ast_format *format1, const
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/* Use the maximum packetloss percentage between the two attributes. This affects how
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* much redundancy is used in the FEC. */
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attr_res->packetloss_percentage = MAX(attr1->packetloss_percentage, attr2->packetloss_percentage);
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}
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return jointformat;
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}
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@@ -183,9 +189,7 @@ static struct ast_format *silk_set(const struct ast_format *format, const char *
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}
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attr = ast_format_get_attribute_data(cloned);
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if (!strcasecmp(name, "sample_rate")) {
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attr->samplerate = val;
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} else if (!strcasecmp(name, "max_bitrate")) {
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if (!strcasecmp(name, "max_bitrate")) {
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attr->maxbitrate = val;
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} else if (!strcasecmp(name, "dtx")) {
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attr->dtx = val;
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@@ -205,9 +209,7 @@ static const void *silk_get(const struct ast_format *format, const char *name)
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struct silk_attr *attr = ast_format_get_attribute_data(format);
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unsigned int *val;
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if (!strcasecmp(name, "sample_rate")) {
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val = &attr->samplerate;
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} else if (!strcasecmp(name, "max_bitrate")) {
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if (!strcasecmp(name, "max_bitrate")) {
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val = &attr->maxbitrate;
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} else if (!strcasecmp(name, "dtx")) {
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val = &attr->dtx;
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