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rtspsrc: Keep track of connected state

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Keep track of the state of the connection and don't try to send TEARDOWN when
the server has closed the connection.
This commit is contained in:
Wim Taymans 2009-02-04 11:38:30 +01:00
parent 7353cdc029
commit 1dc5c34143
2 changed files with 15 additions and 11 deletions
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25
gst/rtsp/gstrtspsrc.c
View file

@ -2911,6 +2911,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
gst_rtsp_connection_connect (src->connection, src->ptcp_timeout);
if (res < 0)
goto connect_error;
src->connected = TRUE;
continue;
default:
goto receive_error;
@ -3025,6 +3026,7 @@ connect_error:
{
gchar *str = gst_rtsp_strresult (res);
src->connected = FALSE;
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not connect to server. (%s)", str));
g_free (str);
@ -3471,6 +3473,8 @@ receive_error:
gst_rtsp_connection_connect (src->connection,
src->ptcp_timeout)) == 0)
goto again;
src->connected = FALSE;
}
/* only try once after reconnect, then fallthrough and error out */
default:
@ -3978,8 +3982,10 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src)
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
&resptrans, 0);
if (!resptrans)
if (!resptrans) {
gst_rtspsrc_stream_free_udp (stream);
goto no_transport;
}
/* parse transport, go to next stream on parse error */
if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
@ -4191,6 +4197,8 @@ restart:
gst_rtsp_connection_connect (src->connection, src->ptcp_timeout)) < 0)
goto could_not_connect;
src->connected = TRUE;
/* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options...");
res =
@ -4238,6 +4246,7 @@ restart:
gst_rtsp_connection_close (src->connection);
gst_rtsp_connection_free (src->connection);
src->connection = NULL;
src->connected = FALSE;
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
@ -4375,6 +4384,7 @@ cleanup_error:
GST_DEBUG_OBJECT (src, "free connection");
gst_rtsp_connection_free (src->connection);
src->connection = NULL;
src->connected = FALSE;
}
GST_RTSP_STATE_UNLOCK (src);
gst_rtsp_message_unset (&request);
@ -4436,7 +4446,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
GST_DEBUG_OBJECT (src, "stop connection flush");
gst_rtsp_connection_flush (src->connection, FALSE);
if (src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)) {
if (src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN) && src->connected) {
/* do TEARDOWN */
res =
gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN,
@ -4457,12 +4467,11 @@ gst_rtspsrc_close (GstRTSPSrc * src)
/* close connection */
GST_DEBUG_OBJECT (src, "closing connection...");
if ((res = gst_rtsp_connection_close (src->connection)) < 0)
goto close_failed;
gst_rtsp_connection_close (src->connection);
/* free connection */
gst_rtsp_connection_free (src->connection);
src->connection = NULL;
src->connected = FALSE;
done:
/* cleanup */
@ -4489,12 +4498,6 @@ send_error:
("Could not send message."));
return FALSE;
}
close_failed:
{
GST_RTSP_STATE_UNLOCK (src);
GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, (NULL), ("Close failed."));
return FALSE;
}
}
/* RTP-Info is of the format:

1
gst/rtsp/gstrtspsrc.h
View file

@ -204,6 +204,7 @@ struct _GstRTSPSrc {
gulong session_ptmap_id;
GstRTSPConnection *connection;
gboolean connected;
/* a list of RTSP extensions as GstElement */
GstRTSPExtensionList *extensions;