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/1.4@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index f8a929d6e505aeed009a7d6f0869011be50d71ab..167db198deae9cfd20eebe61532fc8ae34745083 100644 (file)
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -918,7 +918,7 @@ 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 3f8b49e0dda36cb072df1de1d136850bb621a99e..8ac448a8c0a292b2383eef5dd79a12c4c2cbc7dd 100644 (file)
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1442,7 +1442,7 @@ 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 6d80d466f38818540d8ea3393cbff4581ad11036..831f433a3636765cbcb69bfc0c894ce8f3171712 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3864,7 +3864,7 @@ static int sip_answer(struct ast_channel *ast)
                if (option_debug)
                        ast_log(LOG_DEBUG, "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);
                ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
        }
@@ -3899,7 +3899,7 @@ 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, 1);
@@ -4147,11 +4147,11 @@ 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 */
@@ -4162,7 +4162,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
                        res = -1;
                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 48be77b9c3d7cfd8ced8e26d976c8163c36dffac..59e8cfa790da893d0c897ef6c6fa906a6b6f2575 100644 (file)
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -2868,7 +2868,7 @@ 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 955e8bf99ca0ca45e16f34fb2e549c49339460a2..14e4b83967519e8c4e0e246184d87b58d99622fa 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -180,7 +180,7 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
 int ast_rtp_settos(struct ast_rtp *rtp, int tos);
 
 /*! \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 76deb3c693b95bc1a37750b4dd1090c0945b9a97..26127d2a432bcaee9451fc28d971f1d2fb5ce92d 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -2082,7 +2082,7 @@ int ast_rtp_settos(struct ast_rtp *rtp, int tos)
        return res;
 }
 
-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;