[asterisk-commits] oej: trunk r101268 - in /trunk: include/asterisk/rtp.h main/rtp.c

[SVN commits to the Asterisk project]

Author: oej
Date: Wed Jan 30 10:39:14 2008
New Revision: 101268

URL: http://svn.digium.com/view/asterisk?view=rev&rev=101268
Log:
- doxygen fixes
- change function to void because it always returned the same value and no one read it.

Modified:
    trunk/include/asterisk/rtp.h
    trunk/main/rtp.c

Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?view=diff&rev=101268&r1=101267&r2=101268
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Wed Jan 30 10:39:14 2008
@@ -262,13 +262,13 @@
 int ast_rtp_reload(void);                                     /*! reload rtp configuration */
 void ast_rtp_new_init(struct ast_rtp *rtp);
 
-/*! Set codec preference */
-int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
-
-/*! Get codec preference */
+/*! \brief Set codec preference */
+void ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
+
+/*! \brief Get codec preference */
 struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
 
-/*! get format from predefined dynamic payload format */
+/*! \brief get format from predefined dynamic payload format */
 int ast_rtp_codec_getformat(int pt);
 
 /*! \brief Set rtp timeout */

Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=101268&r1=101267&r2=101268
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Wed Jan 30 10:39:14 2008
@@ -1798,7 +1798,6 @@
 
 int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
 {
-	// dest = c0, src = c1
 	struct ast_rtp *destp = NULL, *srcp = NULL;		/* Audio RTP Channels */
 	struct ast_rtp *vdestp = NULL, *vsrcp = NULL;		/* Video RTP channels */
 	struct ast_rtp *tdestp = NULL, *tsrcp = NULL;		/* Text RTP channels */
@@ -2034,8 +2033,6 @@
 	}
 
 	rtp_bridge_unlock(rtp);
-	
-	return;
 }
 
 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt) 
@@ -2217,8 +2214,6 @@
 	rtp->seqno = ast_random() & 0xffff;
 	ast_set_flag(rtp, FLAG_HAS_DTMF);
 	rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
-
-	return;
 }
 
 struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr addr)
@@ -3086,7 +3081,7 @@
 	return 0;
 }
 
-int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs)
+void ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs)
 {
 	int x;
 	for (x = 0; x < 32; x++) {  /* Ugly way */
@@ -3096,7 +3091,6 @@
 	if (rtp->smoother)
 		ast_smoother_free(rtp->smoother);
 	rtp->smoother = NULL;
-	return 0;
 }
 
 struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp)
@@ -3518,8 +3512,6 @@
 	rtp_bridge_lock(rtp0);
 	rtp0->bridged = rtp1;
 	rtp_bridge_unlock(rtp0);
-
-	return;
 }
 
 /*! \brief Bridge loop for partial native bridge (packet2packet)