adds support for slin16 in sip
(closes issue #16153) Reported by: kfister Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested by: kfister, malcolmd git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271261 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -68,6 +68,7 @@ SIP Changes
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* Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
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Charge messages to snom phones.
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* Added support for G.719 media streams.
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* Added support for 16khz signed linear media streams.
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IAX2 Changes
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-----------
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@ -97,6 +97,7 @@ static const struct ast_rtp_mime_type {
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{{1, AST_FORMAT_G726}, "audio", "G726-32", 8000},
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{{1, AST_FORMAT_ADPCM}, "audio", "DVI4", 8000},
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{{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
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{{1, AST_FORMAT_SLINEAR16}, "audio", "L16", 16000},
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{{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
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{{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
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{{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
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@ -165,15 +166,16 @@ static const struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] = {
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[102] = {1, AST_FORMAT_SIREN7},
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[103] = {1, AST_FORMAT_H263_PLUS},
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[104] = {1, AST_FORMAT_MP4_VIDEO},
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[105] = {1, AST_FORMAT_T140RED}, /* Real time text chat (with redundancy encoding) */
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[106] = {1, AST_FORMAT_T140}, /* Real time text chat */
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[105] = {1, AST_FORMAT_T140RED}, /* Real time text chat (with redundancy encoding) */
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[106] = {1, AST_FORMAT_T140}, /* Real time text chat */
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[110] = {1, AST_FORMAT_SPEEX},
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[111] = {1, AST_FORMAT_G726},
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[112] = {1, AST_FORMAT_G726_AAL2},
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[115] = {1, AST_FORMAT_SIREN14},
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[116] = {1, AST_FORMAT_G719},
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[117] = {1, AST_FORMAT_SPEEX16},
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[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
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[118] = {1, AST_FORMAT_SLINEAR16}, /* 16 Khz signed linear */
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[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
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};
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int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
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@ -2230,7 +2230,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
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if (rtp->f.subclass.codec & AST_FORMAT_AUDIO_MASK) {
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rtp->f.samples = ast_codec_get_samples(&rtp->f);
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if (rtp->f.subclass.codec == AST_FORMAT_SLINEAR)
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if (rtp->f.subclass.codec == AST_FORMAT_SLINEAR || AST_FORMAT_SLINEAR16)
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ast_frame_byteswap_be(&rtp->f);
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calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
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/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
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