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/1.6.1@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 32ba5060edb6a8a5d679a83a48a2e72f16950c5e..18a6c8337f466f7cfa53003f19d6dd40bf6cdad2 100644 (file)
--- 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 f30ddd15c669131e3e774477d6960a3b08887f62..a078b80459e6b7f7765ab71f2eeb6629e6d6d2e1 100644 (file)
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1480,7 +1480,7 @@ 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 b97c8cb9a188e8f191952b25ce45003437a1ddfa..5fb2d22836add301442c6c27d2c116dff1f6ef28 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5785,7 +5785,7 @@ 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);
        }
@@ -5820,7 +5820,7 @@ 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);
@@ -6143,11 +6143,11 @@ 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 */
@@ -6166,7 +6166,7 @@ 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 9839ee2ebbe9151463cff26644d945fcf19f7fc4..9bffdb7fc7a3b6dcdfe7f68b876bc08c74a624f1 100644 (file)
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -3918,7 +3918,7 @@ 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 07491024a37a355e3283ab703a91de49ac991bfd..4ddd373f49da8a599404c20b8811f572040db640 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -211,7 +211,7 @@ 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 5b1cc57c3bd31e150dea4725965c9a44fd124738..34806441d4f65b2be6eb7175fb566e396aa35776 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -2616,7 +2616,7 @@ 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;