[asterisk-commits] file: branch 1.4 r70360 - in /branches/1.4/main: frame.c rtp.c

[SVN commits to the Asterisk project]

Author: file
Date: Wed Jun 20 12:52:57 2007
New Revision: 70360

URL: http://svn.digium.com/view/asterisk?view=rev&rev=70360
Log:
Put the speex packetization values back in but disable it when setting up the smoother.

Modified:
    branches/1.4/main/frame.c
    branches/1.4/main/rtp.c

Modified: branches/1.4/main/frame.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/frame.c?view=diff&rev=70360&r1=70359&r2=70360
==============================================================================
--- branches/1.4/main/frame.c (original)
+++ branches/1.4/main/frame.c Wed Jun 20 12:52:57 2007
@@ -113,7 +113,7 @@
 	{ 1, AST_FORMAT_SLINEAR, "slin", "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE },	/*!< 7 */
 	{ 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 },		/*!<  8: codec_lpc10.c */ 
 	{ 1, AST_FORMAT_G729A, "g729", "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 },	/*!<  9: Binary commercial distribution */
-	{ 1, AST_FORMAT_SPEEX, "speex", "SpeeX"},		/*!< 10: codec_speex.c */
+	{ 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 },		/*!< 10: codec_speex.c */
 	{ 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 },		/*!< 11: codec_ilbc.c */ /* inc=30ms - workaround */
 	{ 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 300, 10, 20 },	/*!<  12: codec_g726.c */
 	{ 1, AST_FORMAT_G722, "g722", "G722"},					/*!< 13 */

Modified: branches/1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/rtp.c?view=diff&rev=70360&r1=70359&r2=70360
==============================================================================
--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Wed Jun 20 12:52:57 2007
@@ -2728,7 +2728,7 @@
 		rtp->smoother = NULL;
 	}
 
-	if (!rtp->smoother) {
+	if (!rtp->smoother && subclass != AST_FORMAT_SPEEX) {
 		struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
 		if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
 			if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {