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: 1.6.0.27-rc1~37
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=5954201d07f1746e9ba79d995f082a19992bae5f;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/1.6.0@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 6408367315..8c4d734e4b 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -919,7 +919,7 @@ 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 1e02db100b..2eb0457893 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1477,7 +1477,7 @@ 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 e861f127b8..9878a2c31c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5315,7 +5315,7 @@ 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);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
 	}
@@ -5350,7 +5350,7 @@ 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);
@@ -5670,11 +5670,11 @@ 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 */
@@ -5693,7 +5693,7 @@ 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 12533fbfc8..17d31202a9 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -3769,7 +3769,7 @@ 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 696fe7a950..8a4ec0f517 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -188,7 +188,7 @@ 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 8d6269397a..4646dd24d6 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -2401,7 +2401,7 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc)
 	return ast_netsock_set_qos(rtp->s, tos, cos, 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;