Commit 9e37243e authored by Wim Taymans's avatar Wim Taymans
Browse files

gst/rtsp/gstrtspsrc.c: Let more error state trickle down so that we can catch more error cases.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
Let more error state trickle down so that we can catch more error
cases.
Handle keep-alive a little smarter by selecting a method the server
actually supports.
Fix a race in UDP streaming shutdown.
parent 5f2fbbd7
2007-05-04 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
Let more error state trickle down so that we can catch more error
cases.
Handle keep-alive a little smarter by selecting a method the server
actually supports.
Fix a race in UDP streaming shutdown.
2007-05-04 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
......
......@@ -2043,17 +2043,22 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
RTSPMessage response = { 0 };
RTSPResult res;
RTSPStatusCode code;
RTSPMethod method;
GST_DEBUG_OBJECT (src, "creating server keep-alive");
res =
rtsp_message_init_request (&request, RTSP_GET_PARAMETER,
src->req_location);
/* find a method to use for keep-alive */
if (src->methods & RTSP_GET_PARAMETER)
method = RTSP_GET_PARAMETER;
else
method = RTSP_OPTIONS;
res = rtsp_message_init_request (&request, method, src->req_location);
if (res < 0)
goto send_error;
/* let us handle the error code because we don't care */
if (!gst_rtspsrc_send (src, &request, &response, &code))
if ((res = gst_rtspsrc_send (src, &request, &response, &code)) < 0)
goto send_error;
rtsp_message_unset (&request);
......@@ -2224,6 +2229,11 @@ receive_error:
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
/* don't bother continueing if we the connection was closed */
if (res == RTSP_EEOF) {
src->running = FALSE;
gst_task_pause (src->task);
}
return;
}
handle_request_failed:
......@@ -2419,7 +2429,7 @@ no_user_pass:
}
}
static gboolean
static RTSPResult
gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
......@@ -2463,13 +2473,15 @@ next:
thecode = response->type_data.response.code;
GST_DEBUG_OBJECT (src, "got response message %d", thecode);
/* if the caller wanted the result code, we store it. */
if (code)
*code = thecode;
/* If the request didn't succeed, bail out before doing any more */
if (thecode != RTSP_STS_OK)
return FALSE;
return RTSP_OK;
/* store new content base if any */
rtsp_message_get_header (response, RTSP_HDR_CONTENT_BASE, &content_base);
......@@ -2479,7 +2491,7 @@ next:
if (src->extension && src->extension->after_send)
src->extension->after_send (src->extension, request, response);
return TRUE;
return RTSP_OK;
/* ERRORS */
send_error:
......@@ -2489,7 +2501,7 @@ send_error:
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
g_free (str);
return FALSE;
return res;
}
receive_error:
{
......@@ -2498,12 +2510,12 @@ receive_error:
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
return FALSE;
return res;
}
handle_request_failed:
{
/* ERROR was posted */
return FALSE;
return res;
}
}
......@@ -2526,19 +2538,20 @@ handle_request_failed:
* to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
* the request.
*
* Returns: TRUE if the processing was successful.
* Returns: RTSP_OK if the processing was successful.
*/
gboolean
RTSPResult
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPStatusCode int_code = RTSP_STS_OK;
gboolean res;
RTSPResult res;
gboolean retry;
do {
retry = FALSE;
res = gst_rtspsrc_try_send (src, request, response, &int_code);
if ((res = gst_rtspsrc_try_send (src, request, response, &int_code)) < 0)
goto error;
if (int_code == RTSP_STS_UNAUTHORIZED) {
if (gst_rtspsrc_setup_auth (src, response)) {
......@@ -2558,6 +2571,12 @@ gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
return res;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (src, "got error %d", res);
return res;
}
error_response:
{
switch (response->type_data.response.code) {
......@@ -2573,7 +2592,7 @@ error_response:
}
/* we return FALSE so we should unset the response ourselves */
rtsp_message_unset (response);
return FALSE;
return RTSP_ERROR;
}
}
......@@ -2872,7 +2891,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src)
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
g_free (transports);
if (!gst_rtspsrc_send (src, &request, &response, NULL))
if ((res = gst_rtspsrc_send (src, &request, &response, NULL) < 0))
goto send_error;
/* parse response transport */
......@@ -3019,7 +3038,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* send OPTIONS */
GST_DEBUG_OBJECT (src, "send options...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0)
goto send_error;
/* parse OPTIONS */
......@@ -3043,7 +3062,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* send DESCRIBE */
GST_DEBUG_OBJECT (src, "send describe...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0)
goto send_error;
/* check if reply is SDP */
......@@ -3219,7 +3238,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0)
goto send_error;
/* FIXME, parse result? */
......@@ -3375,7 +3394,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0-");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0)
goto send_error;
rtsp_message_unset (&request);
......@@ -3466,7 +3485,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0)
goto send_error;
rtsp_message_unset (&request);
......@@ -3621,7 +3640,7 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (rtspsrc, "start flush");
rtsp_connection_flush (rtspsrc->connection, TRUE);
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_STOP, TRUE);
break;
default:
break;
......
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