Revert API change in release branches

This re-renames ast_rtp_update_source to ast_rtp_new_source


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_h323.c

@@ -914,7 +914,7 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data
 		res = 0;
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_update_source(pvt->rtp);
+		ast_rtp_new_source(pvt->rtp);
 		res = 0;
 		break;
 	case AST_CONTROL_SRCCHANGE:

channels/chan_mgcp.c

@@ -1454,7 +1454,7 @@ static int mgcp_indicate(struct ast_channel *ast, int ind, const void *data, siz
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_update_source(sub->rtp);
+		ast_rtp_new_source(sub->rtp);
 		break;
 	case AST_CONTROL_SRCCHANGE:
 		ast_rtp_change_source(sub->rtp);

channels/chan_sip.c

@@ -6188,7 +6188,7 @@ static int sip_answer(struct ast_channel *ast)
 
 		ast_setstate(ast, AST_STATE_UP);
 		ast_debug(1, "SIP answering channel: %s\n", ast->name);
-		ast_rtp_update_source(p->rtp);
+		ast_rtp_new_source(p->rtp);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 	}
@@ -6223,7 +6223,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
 				if ((ast->_state != AST_STATE_UP) &&
 				    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
 				    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
-					ast_rtp_update_source(p->rtp);
+					ast_rtp_new_source(p->rtp);
 					if (!global_prematuremediafilter) {
 						p->invitestate = INV_EARLY_MEDIA;
 						transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
@@ -6546,11 +6546,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 		res = -1;
 		break;
 	case AST_CONTROL_HOLD:
-		ast_rtp_update_source(p->rtp);
+		ast_rtp_new_source(p->rtp);
 		ast_moh_start(ast, data, p->mohinterpret);
 		break;
 	case AST_CONTROL_UNHOLD:
-		ast_rtp_update_source(p->rtp);
+		ast_rtp_new_source(p->rtp);
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_VIDUPDATE:	/* Request a video frame update */
@@ -6569,7 +6569,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 		}
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_update_source(p->rtp);
+		ast_rtp_new_source(p->rtp);
 		break;
 	case AST_CONTROL_SRCCHANGE:
 		ast_rtp_change_source(p->rtp);

channels/chan_skinny.c

@@ -4256,7 +4256,7 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
 	case AST_CONTROL_PROCEEDING:
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_update_source(sub->rtp);
+		ast_rtp_new_source(sub->rtp);
 		break;
 	case AST_CONTROL_SRCCHANGE:
 		ast_rtp_change_source(sub->rtp);

include/asterisk/rtp.h

@@ -217,7 +217,7 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
 int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
 
 /*! \brief Indicate that we need to set the marker bit */
-void ast_rtp_update_source(struct ast_rtp *rtp);
+void ast_rtp_new_source(struct ast_rtp *rtp);
 
 /*! \brief Indicate that we need to set the marker bit and change the ssrc */
 void ast_rtp_change_source(struct ast_rtp *rtp);

main/rtp.c

@@ -2657,7 +2657,7 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int type_of_service, int class_of_servic
 	return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
 }
 
-void ast_rtp_update_source(struct ast_rtp *rtp)
+void ast_rtp_new_source(struct ast_rtp *rtp)
 {
 	if (rtp) {
 		rtp->set_marker_bit = 1;