From: David Vossel <dvossel@digium.com>
Date: Thu, 17 Jun 2010 18:36:06 +0000 (+0000)
Subject: adds support for slin16 in sip
X-Git-Tag: 11.0.0-beta1~2817
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=ba3d1ad680ca96981537cebeb97a6045f593d033;p=thirdparty%2Fasterisk.git

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: https://origsvn.digium.com/svn/asterisk/trunk@271261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/CHANGES b/CHANGES
index 779adfbbdc..d59eebd118 100644
--- a/CHANGES
+++ b/CHANGES
@@ -68,6 +68,7 @@ SIP Changes
  * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
    Charge messages to snom phones.
  * Added support for G.719 media streams.
+ * Added support for 16khz signed linear media streams.
 
 IAX2 Changes
 -----------
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 5e4db23094..eeaa008a8b 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -97,6 +97,7 @@ static const struct ast_rtp_mime_type {
 	{{1, AST_FORMAT_G726}, "audio", "G726-32", 8000},
 	{{1, AST_FORMAT_ADPCM}, "audio", "DVI4", 8000},
 	{{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
+	{{1, AST_FORMAT_SLINEAR16}, "audio", "L16", 16000},
 	{{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
 	{{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
 	{{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
@@ -165,15 +166,16 @@ static const struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] = {
 	[102] = {1, AST_FORMAT_SIREN7},
 	[103] = {1, AST_FORMAT_H263_PLUS},
 	[104] = {1, AST_FORMAT_MP4_VIDEO},
-	[105] = {1, AST_FORMAT_T140RED},        /* Real time text chat (with redundancy encoding) */
-	[106] = {1, AST_FORMAT_T140},   /* Real time text chat */
+	[105] = {1, AST_FORMAT_T140RED},   /* Real time text chat (with redundancy encoding) */
+	[106] = {1, AST_FORMAT_T140},      /* Real time text chat */
 	[110] = {1, AST_FORMAT_SPEEX},
 	[111] = {1, AST_FORMAT_G726},
 	[112] = {1, AST_FORMAT_G726_AAL2},
 	[115] = {1, AST_FORMAT_SIREN14},
 	[116] = {1, AST_FORMAT_G719},
 	[117] = {1, AST_FORMAT_SPEEX16},
-	[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
+	[118] = {1, AST_FORMAT_SLINEAR16}, /* 16 Khz signed linear */
+	[121] = {0, AST_RTP_CISCO_DTMF},   /* Must be type 121 */
 };
 
 int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 7fe8ff9ba5..edea6112c0 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -2230,7 +2230,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 
 	if (rtp->f.subclass.codec & AST_FORMAT_AUDIO_MASK) {
 		rtp->f.samples = ast_codec_get_samples(&rtp->f);
-		if (rtp->f.subclass.codec == AST_FORMAT_SLINEAR)
+		if (rtp->f.subclass.codec == AST_FORMAT_SLINEAR || AST_FORMAT_SLINEAR16)
 			ast_frame_byteswap_be(&rtp->f);
 		calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
 		/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */