From: Terry Wilson <twilson@digium.com>
Date: Wed, 17 Mar 2010 16:25:52 +0000 (+0000)
Subject: Revert API change in release branches
X-Git-Tag: 1.6.2.7-rc1~38
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=dbf0243679ba00f5801d53964287ed6ea91beb81;p=thirdparty%2Fasterisk.git

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
---

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 5734fbb584..720236cb92 100644
--- a/channels/chan_h323.c
+++ b/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:
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 319ce91012..787d45d16e 100644
--- a/channels/chan_mgcp.c
+++ b/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);
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d4ff2f788b..31c910a886 100644
--- a/channels/chan_sip.c
+++ b/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);
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index c7341376a8..dd0f7466df 100644
--- a/channels/chan_skinny.c
+++ b/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);
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index e7c80c58a5..aa38ee0fa1 100644
--- a/include/asterisk/rtp.h
+++ b/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);
diff --git a/main/rtp.c b/main/rtp.c
index 4f88468028..9d46f97204 100644
--- a/main/rtp.c
+++ b/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;