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rtspsrc: fix memory leak in close

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Close the connection even when we fail to send the teardown message.
Use the connection url (which is a copy of the src url).
This commit is contained in:
Wim Taymans 2009-03-04 16:25:34 +01:00
parent c83635d42c
commit 5a5ba49c9b
1 changed files with 22 additions and 9 deletions
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31
gst/rtsp/gstrtspsrc.c
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@ -3334,6 +3334,7 @@ gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
GstRTSPAuthMethod method;
GstRTSPResult auth_result;
GstRTSPUrl *url;
gchar *hdr;
/* Identify the available auth methods and see if any are supported */
@ -3349,10 +3350,12 @@ gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
* data are stale, we just update them in the connection object and
* return TRUE to retry the request */
url = gst_rtsp_connection_get_url (src->connection);
/* Do we have username and password available? */
if (src->url != NULL && !src->tried_url_auth) {
user = src->url->user;
pass = src->url->passwd;
if (url != NULL && !src->tried_url_auth) {
user = url->user;
pass = url->passwd;
src->tried_url_auth = TRUE;
GST_DEBUG_OBJECT (src,
"Attempting authentication using credentials from the URL");
@ -3909,10 +3912,13 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src)
GstRTSPLowerTrans protocols;
GstRTSPStatusCode code;
gint rtpport, rtcpport;
GstRTSPUrl *url;
url = gst_rtsp_connection_get_url (src->connection);
/* we initially allow all configured lower transports. based on the URL
* transports and the replies from the server we narrow them down. */
protocols = src->url->transports & src->cur_protocols;
protocols = url->transports & src->cur_protocols;
if (protocols == 0)
goto no_protocols;
@ -4095,7 +4101,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src)
}
}
gst_rtsp_ext_list_stream_select (src->extensions, src->url);
gst_rtsp_ext_list_stream_select (src->extensions, url);
/* if there is nothing to activate, error out */
if (!src->need_activate)
@ -4223,6 +4229,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
GstSDPMessage sdp = { 0 };
GstRTSPStream *stream = NULL;
gchar *respcont = NULL;
GstRTSPUrl *url;
GST_RTSP_STATE_LOCK (src);
@ -4242,7 +4249,9 @@ restart:
if ((res = gst_rtsp_connection_create (src->url, &src->connection)) < 0)
goto could_not_create;
if (src->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
url = gst_rtsp_connection_get_url (src->connection);
if (url->transports & GST_RTSP_LOWER_TRANS_HTTP)
gst_rtsp_connection_set_tunneled (src->connection, TRUE);
/* connect */
@ -4471,6 +4480,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res;
gboolean ret = FALSE;
GST_DEBUG_OBJECT (src, "TEARDOWN...");
@ -4519,6 +4529,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
"TEARDOWN and PLAY not supported, can't do TEARDOWN");
}
close:
/* close connection */
GST_DEBUG_OBJECT (src, "closing connection...");
gst_rtsp_connection_close (src->connection);
@ -4534,7 +4545,7 @@ done:
src->state = GST_RTSP_STATE_INVALID;
GST_RTSP_STATE_UNLOCK (src);
return TRUE;
return ret;
/* ERRORS */
create_request_failed:
@ -4542,7 +4553,8 @@ create_request_failed:
GST_RTSP_STATE_UNLOCK (src);
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request."));
return FALSE;
ret = FALSE;
goto close;
}
send_error:
{
@ -4550,7 +4562,8 @@ send_error:
gst_rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message."));
return FALSE;
ret = FALSE;
goto close;
}
}