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 5734fbb5846b66d3c7314ccce845e1574b20afad..720236cb921bd48d5ea42148a8a6d4aef8551b53 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 319ce910121067ba9cb126d281760f01aafe85ab..787d45d16ecb0d51f40ec2c11b06c3a5c1be00f1 100644 (file)
--- 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 d4ff2f788b3156f550ff93dfb17887e7c146df9b..31c910a886cff8b582a12068a7598471cfaf0fca 100644 (file)
--- 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 c7341376a831d60937a605085f970d7d5474b6e9..dd0f7466df3c1780bee635e9ff43567da34c6116 100644 (file)
--- 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 e7c80c58a5c9b3cb004b43eeb9ba538d79e6bad0..aa38ee0fa1a11d5c3058e1725d7ffa583e58595c 100644 (file)
--- 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 4f88468028ae103cdd2530e716a3125dfb686a15..9d46f97204306944609bc51472cfc8be3318fe92 100644 (file)
--- 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;