[asterisk-commits] oej: branch oej/videocaps r47890 - in /team/oej/videocaps: channels/ main/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Tue Nov 21 09:12:25 2006
New Revision: 47890

URL: http://svn.digium.com/view/asterisk?view=rev&rev=47890
Log:
Compilation errors now fixed

Modified:
    team/oej/videocaps/channels/chan_sip.c
    team/oej/videocaps/main/rtp.c

Modified: team/oej/videocaps/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_sip.c?view=diff&rev=47890&r1=47889&r2=47890
==============================================================================
--- team/oej/videocaps/channels/chan_sip.c (original)
+++ team/oej/videocaps/channels/chan_sip.c Tue Nov 21 09:12:25 2006
@@ -2911,14 +2911,15 @@
 	struct ast_var_t *current;
 	const char *referer = NULL;   /* SIP refererer */	
 
-  if(sipdebug_caps)
-		ast_verbose("SIPCALL: entering with channel->nativeformats=%08x, p->capability=%08x, p->jointcaps=%08x\n", ast->nativeformats, p->capability, p->jointcapability);
 	
 	p = ast->tech_pvt;
 	if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
 		ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
 		return -1;
 	}
+
+	if(sipdebug_caps)
+		ast_verbose("SIPCALL: entering with channel->nativeformats=%08x, p->capability=%08x, p->jointcaps=%08x\n", ast->nativeformats, p->capability, p->jointcapability);
 
 	/* Check whether there is vxml_url, distinctive ring variables */
 	headp=&ast->varshead;
@@ -3665,7 +3666,7 @@
 		break;
 	case AST_FRAME_TEXT:
 		if (p) {
-			ast_mutex_lock(&p->lock);
+			sip_pvt_lock(p);
 			if (p->trtp) {
 				/* Activate text early media */
 				if ((ast->_state != AST_STATE_UP) &&
@@ -3677,7 +3678,7 @@
 				p->lastrtptx = time(NULL);
 				res = ast_rtp_write(p->trtp, frame);
 			}
-			ast_mutex_unlock(&p->lock);
+			sip_pvt_unlock(p);
 		}
 		break;
 	case AST_FRAME_IMAGE:
@@ -17726,9 +17727,9 @@
 	if (!(p = chan->tech_pvt))
 		return AST_RTP_GET_FAILED;
 
-	ast_mutex_lock(&p->lock);
+	sip_pvt_lock(p);
 	if (!(p->trtp)) {
-		ast_mutex_unlock(&p->lock);
+		sip_pvt_unlock(p);
 		return AST_RTP_GET_FAILED;
 	}
 
@@ -17737,7 +17738,7 @@
 	if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
 		res = AST_RTP_TRY_NATIVE;
 
-	ast_mutex_unlock(&p->lock);
+	sip_pvt_unlock(p);
 
 	return res;
 }

Modified: team/oej/videocaps/main/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/main/rtp.c?view=diff&rev=47890&r1=47889&r2=47890
==============================================================================
--- team/oej/videocaps/main/rtp.c (original)
+++ team/oej/videocaps/main/rtp.c Tue Nov 21 09:12:25 2006
@@ -1509,11 +1509,11 @@
 	/* Get audio, video  and text interface (if native bridge is possible) */
 	audio_dest_res = destpr->get_rtp_info(c0, &destp);
 	video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(c0, &vdestp) : AST_RTP_GET_FAILED;
-	text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(dest, &tdestp) : AST_RTP_GET_FAILED;
+	text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(c0, &tdestp) : AST_RTP_GET_FAILED;
 	if (srcpr) {
 		audio_src_res = srcpr->get_rtp_info(c1, &srcp);
 		video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(c1, &vsrcp) : AST_RTP_GET_FAILED;
-		text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(src, &tscrcp) : AST_RTP_GET_FAILED;
+		text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED;
 	}
 
 	/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
@@ -1534,7 +1534,7 @@
 	if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
 		nat_active = 1;
 	/* Bridge media early */
-	if (destpr->set_rtp_peer(c0, srcp, vsrcp, tshp, srccodec, nat_active))
+	if (destpr->set_rtp_peer(c0, srcp, vsrcp, tsrcp, srccodec, nat_active))
 		ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
 	ast_channel_unlock(c0);
 	if (c1)