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

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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.
This commit is contained in:
Wim Taymans 2007-05-04 15:17:14 +00:00
parent 5f2fbbd76b
commit 9e37243eca
2 changed files with 54 additions and 22 deletions
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13
ChangeLog
View file

@ -1,3 +1,16 @@
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> 2007-05-04 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):

63
gst/rtsp/gstrtspsrc.c
View file

@ -2043,17 +2043,22 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
RTSPMessage response = { 0 }; RTSPMessage response = { 0 };
RTSPResult res; RTSPResult res;
RTSPStatusCode code; RTSPStatusCode code;
RTSPMethod method;
GST_DEBUG_OBJECT (src, "creating server keep-alive"); GST_DEBUG_OBJECT (src, "creating server keep-alive");
res = /* find a method to use for keep-alive */
rtsp_message_init_request (&request, RTSP_GET_PARAMETER, if (src->methods & RTSP_GET_PARAMETER)
src->req_location); method = RTSP_GET_PARAMETER;
else
method = RTSP_OPTIONS;
res = rtsp_message_init_request (&request, method, src->req_location);
if (res < 0) if (res < 0)
goto send_error; goto send_error;
/* let us handle the error code because we don't care */ /* 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; goto send_error;
rtsp_message_unset (&request); rtsp_message_unset (&request);
@ -2224,6 +2229,11 @@ receive_error:
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL), GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str)); ("Could not receive message. (%s)", str));
g_free (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; return;
} }
handle_request_failed: handle_request_failed:
@ -2419,7 +2429,7 @@ no_user_pass:
} }
} }
static gboolean static RTSPResult
gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request, gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code) RTSPMessage * response, RTSPStatusCode * code)
{ {
@ -2463,13 +2473,15 @@ next:
thecode = response->type_data.response.code; 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 the caller wanted the result code, we store it. */
if (code) if (code)
*code = thecode; *code = thecode;
/* If the request didn't succeed, bail out before doing any more */ /* If the request didn't succeed, bail out before doing any more */
if (thecode != RTSP_STS_OK) if (thecode != RTSP_STS_OK)
return FALSE; return RTSP_OK;
/* store new content base if any */ /* store new content base if any */
rtsp_message_get_header (response, RTSP_HDR_CONTENT_BASE, &content_base); rtsp_message_get_header (response, RTSP_HDR_CONTENT_BASE, &content_base);
@ -2479,7 +2491,7 @@ next:
if (src->extension && src->extension->after_send) if (src->extension && src->extension->after_send)
src->extension->after_send (src->extension, request, response); src->extension->after_send (src->extension, request, response);
return TRUE; return RTSP_OK;
/* ERRORS */ /* ERRORS */
send_error: send_error:
@ -2489,7 +2501,7 @@ send_error:
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str)); ("Could not send message. (%s)", str));
g_free (str); g_free (str);
return FALSE; return res;
} }
receive_error: receive_error:
{ {
@ -2498,12 +2510,12 @@ receive_error:
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str)); ("Could not receive message. (%s)", str));
g_free (str); g_free (str);
return FALSE; return res;
} }
handle_request_failed: handle_request_failed:
{ {
/* ERROR was posted */ /* 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 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
* the request. * 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, gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code) RTSPMessage * response, RTSPStatusCode * code)
{ {
RTSPStatusCode int_code = RTSP_STS_OK; RTSPStatusCode int_code = RTSP_STS_OK;
gboolean res; RTSPResult res;
gboolean retry; gboolean retry;
do { do {
retry = FALSE; 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 (int_code == RTSP_STS_UNAUTHORIZED) {
if (gst_rtspsrc_setup_auth (src, response)) { if (gst_rtspsrc_setup_auth (src, response)) {
@ -2558,6 +2571,12 @@ gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
return res; return res;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (src, "got error %d", res);
return res;
}
error_response: error_response:
{ {
switch (response->type_data.response.code) { switch (response->type_data.response.code) {
@ -2573,7 +2592,7 @@ error_response:
} }
/* we return FALSE so we should unset the response ourselves */ /* we return FALSE so we should unset the response ourselves */
rtsp_message_unset (response); 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); rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
g_free (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; goto send_error;
/* parse response transport */ /* parse response transport */
@ -3019,7 +3038,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* send OPTIONS */ /* send OPTIONS */
GST_DEBUG_OBJECT (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; goto send_error;
/* parse OPTIONS */ /* parse OPTIONS */
@ -3043,7 +3062,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* send DESCRIBE */ /* send DESCRIBE */
GST_DEBUG_OBJECT (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; goto send_error;
/* check if reply is SDP */ /* check if reply is SDP */
@ -3219,7 +3238,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
if (res < 0) if (res < 0)
goto create_request_failed; 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; goto send_error;
/* FIXME, parse result? */ /* FIXME, parse result? */
@ -3375,7 +3394,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0-"); 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; goto send_error;
rtsp_message_unset (&request); rtsp_message_unset (&request);
@ -3466,7 +3485,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
if (res < 0) if (res < 0)
goto create_request_failed; 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; goto send_error;
rtsp_message_unset (&request); 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_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY: case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (rtspsrc, "start flush"); GST_DEBUG_OBJECT (rtspsrc, "start flush");
rtsp_connection_flush (rtspsrc->connection, TRUE); gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_STOP, TRUE);
break; break;
default: default:
break; break;