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 6408367315d701d0d4bbeee64a3cc488074a00c0..8c4d734e4b62b10a0f816bf914ddfa89410504ce 100644 (file)
--- 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 1e02db100b95974a234cf96d707296bb1ab1065d..2eb04578938b93d6672cc0abcf71f881d4e32852 100644 (file)
--- 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 e861f127b89b03b1ef46d8a33373c025eddd48a3..9878a2c31c11fc0edc37838c616cf2975539f2a0 100644 (file)
--- 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 12533fbfc89ad582e408aa1e6e6625cdb6e6baa9..17d31202a94dd1647b3f6635ec44c6b3315dd59b 100644 (file)
--- 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 696fe7a9501edfb8624a667b10c0da59c890194d..8a4ec0f51713fc959e91a63e7ff045d7f786054e 100644 (file)
--- 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 8d6269397aeac66fda0b9bf4f05f63592d26a0c2..4646dd24d6c5e665733d226214f2a6f36718c08b 100644 (file)
--- 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;