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gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.

Browse source 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Fix race when multiple udp sources post timeouts, just act on the first
received timeout.
Protect stream list with a recursive lock to fix some races.
Flush connection when we need to do a reconnect or stop.
Make state lock recursive.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close):
Some small cleanups.
This commit is contained in:
Wim Taymans 2007-05-02 19:32:58 +00:00
parent 13ae0cde51
commit 24e51b3c73
4 changed files with 182 additions and 102 deletions
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19
ChangeLog
View file

@ -1,3 +1,22 @@
2007-05-02 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Fix race when multiple udp sources post timeouts, just act on the first
received timeout.
Protect stream list with a recursive lock to fix some races.
Flush connection when we need to do a reconnect or stop.
Make state lock recursive.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close):
Some small cleanups.
2007-05-02 Wim Taymans <wim@fluendo.com>
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),

234
gst/rtsp/gstrtspsrc.c
View file

@ -179,12 +179,6 @@ static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
GstStateChange transition);
static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
RTSPMessage * response);
static gboolean gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code);
static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
static gboolean gst_rtspsrc_play (GstRTSPSrc * src);
static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
@ -301,7 +295,8 @@ gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
#endif
src->extension->src = (gpointer) src;
src->state_lock = g_mutex_new ();
src->state_rec_lock = g_new (GStaticRecMutex, 1);
g_static_rec_mutex_init (src->state_rec_lock);
src->state = RTSP_STATE_INVALID;
}
@ -319,7 +314,8 @@ gst_rtspsrc_finalize (GObject * object)
g_free (rtspsrc->content_base);
rtsp_url_free (rtspsrc->url);
g_free (rtspsrc->addr);
g_mutex_free (rtspsrc->state_lock);
g_static_rec_mutex_free (rtspsrc->state_rec_lock);
g_free (rtspsrc->state_rec_lock);
if (rtspsrc->extension) {
#ifdef WITH_EXT_REAL
@ -1050,9 +1046,11 @@ new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src)
gint id, ssrc, pt;
GList *lstream;
GstRTSPStream *stream;
gboolean all_added;
GST_DEBUG_OBJECT (src, "got new session pad %" GST_PTR_FORMAT, pad);
GST_RTSP_STATE_LOCK (src);
/* find stream */
name = gst_object_get_name (GST_OBJECT_CAST (pad));
if (sscanf (name, "recv_rtp_src_%d_%d_%d", &id, &ssrc, &pt) != 3)
@ -1079,23 +1077,30 @@ new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src)
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
/* check if we added all streams */
all_added = TRUE;
for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
stream = (GstRTSPStream *) lstream->data;
if (!stream->added)
goto done;
if (!stream->added) {
all_added = FALSE;
break;
}
}
GST_RTSP_STATE_UNLOCK (src);
if (all_added) {
GST_DEBUG_OBJECT (src, "We added all streams");
/* when we get here, all stream are added and we can fire the no-more-pads
* signal. */
gst_element_no_more_pads (GST_ELEMENT_CAST (src));
}
done:
return;
/* ERRORS */
unknown_stream:
{
GST_DEBUG_OBJECT (src, "ignoring unknown stream");
GST_RTSP_STATE_UNLOCK (src);
g_free (name);
return;
}
@ -1106,21 +1111,26 @@ request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src)
{
GstRTSPStream *stream;
GList *lstream;
GstCaps *caps;
GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
GST_RTSP_STATE_LOCK (src);
lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (session),
(GCompareFunc) find_stream_by_id);
if (!lstream)
goto unknown_stream;
stream = (GstRTSPStream *) lstream->data;
caps = stream->caps;
GST_RTSP_STATE_UNLOCK (src);
return stream->caps;
return caps;
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream %d", session);
GST_RTSP_STATE_UNLOCK (src);
return NULL;
}
}
@ -1852,7 +1862,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
break;
case RTSP_EINTR:
/* we got interrupted, see what we have to do */
GST_DEBUG_OBJECT (src, "we got interrupted");
GST_DEBUG_OBJECT (src, "we got interrupted, unset flushing");
/* unset flushing so we can do something else */
rtsp_connection_flush (src->connection, FALSE);
goto interrupt;
@ -1992,12 +2002,14 @@ play_failed:
}
static void
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
{
GST_OBJECT_LOCK (src);
src->loop_cmd = cmd;
if (cmd != CMD_WAIT)
if (flush) {
GST_DEBUG_OBJECT (src, "start flush");
rtsp_connection_flush (src->connection, TRUE);
}
GST_OBJECT_UNLOCK (src);
}
@ -2153,76 +2165,6 @@ no_user_pass:
}
}
/**
* gst_rtspsrc_send:
* @src: the rtsp source
* @request: must point to a valid request
* @response: must point to an empty #RTSPMessage
*
* send @request and retrieve the response in @response. optionally @code can be
* non-NULL in which case it will contain the status code of the response.
*
* If This function returns TRUE, @response will contain a valid response
* message that should be cleaned with rtsp_message_unset() after usage.
*
* If @code is NULL, this function will return FALSE (with an invalid @response
* message) if the response code was not 200 (OK).
*
* If the attempt results in an authentication failure, then this will attempt
* to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
* the request.
*
* Returns: TRUE if the processing was successful.
*/
gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPStatusCode int_code = RTSP_STS_OK;
gboolean res;
gboolean retry;
do {
retry = FALSE;
res = gst_rtspsrc_try_send (src, request, response, &int_code);
if (int_code == RTSP_STS_UNAUTHORIZED) {
if (gst_rtspsrc_setup_auth (src, response)) {
/* Try the request/response again after configuring the auth info
* and loop again */
retry = TRUE;
}
}
} while (retry == TRUE);
/* If the user requested the code, let them handle errors, otherwise
* post an error below */
if (code != NULL)
*code = int_code;
else if (int_code != RTSP_STS_OK)
goto error_response;
return res;
error_response:
{
switch (response->type_data.response.code) {
case RTSP_STS_NOT_FOUND:
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
response->type_data.response.reason));
break;
default:
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Got error response: %d (%s).", response->type_data.response.code,
response->type_data.response.reason));
break;
}
/* we return FALSE so we should unset the response ourselves */
rtsp_message_unset (response);
return FALSE;
}
}
static gboolean
gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
@ -2234,6 +2176,8 @@ gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
if (src->extension && src->extension->before_send)
src->extension->before_send (src->extension, request);
GST_DEBUG_OBJECT (src, "sending message");
if (src->debug)
rtsp_message_dump (request);
@ -2309,6 +2253,76 @@ handle_request_failed:
}
}
/**
* gst_rtspsrc_send:
* @src: the rtsp source
* @request: must point to a valid request
* @response: must point to an empty #RTSPMessage
*
* send @request and retrieve the response in @response. optionally @code can be
* non-NULL in which case it will contain the status code of the response.
*
* If This function returns TRUE, @response will contain a valid response
* message that should be cleaned with rtsp_message_unset() after usage.
*
* If @code is NULL, this function will return FALSE (with an invalid @response
* message) if the response code was not 200 (OK).
*
* If the attempt results in an authentication failure, then this will attempt
* to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
* the request.
*
* Returns: TRUE if the processing was successful.
*/
gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPStatusCode int_code = RTSP_STS_OK;
gboolean res;
gboolean retry;
do {
retry = FALSE;
res = gst_rtspsrc_try_send (src, request, response, &int_code);
if (int_code == RTSP_STS_UNAUTHORIZED) {
if (gst_rtspsrc_setup_auth (src, response)) {
/* Try the request/response again after configuring the auth info
* and loop again */
retry = TRUE;
}
}
} while (retry == TRUE);
/* If the user requested the code, let them handle errors, otherwise
* post an error below */
if (code != NULL)
*code = int_code;
else if (int_code != RTSP_STS_OK)
goto error_response;
return res;
error_response:
{
switch (response->type_data.response.code) {
case RTSP_STS_NOT_FOUND:
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
response->type_data.response.reason));
break;
default:
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Got error response: %d (%s).", response->type_data.response.code,
response->type_data.response.reason));
break;
}
/* we return FALSE so we should unset the response ourselves */
rtsp_message_unset (response);
return FALSE;
}
}
/* parse the response and collect all the supported methods. We need this
* information so that we don't try to send an unsupported request to the
* server.
@ -2896,6 +2910,23 @@ cleanup_error:
}
}
#if 0
static gboolean
gst_rtspsrc_async_open (GstRTSPSrc * src)
{
GError *error = NULL;
gboolean res = TRUE;
src->thread =
g_thread_create ((GThreadFunc) gst_rtspsrc_open, src, TRUE, &error);
if (error != NULL) {
GST_ELEMENT_ERROR (src, RESOURCE, INIT, (NULL),
("Could not start async thread (%s).", error->message));
}
return res;
}
#endif
static gboolean
gst_rtspsrc_close (GstRTSPSrc * src)
{
@ -2907,15 +2938,15 @@ gst_rtspsrc_close (GstRTSPSrc * src)
GST_RTSP_STATE_LOCK (src);
gst_rtspsrc_loop_send_cmd (src, CMD_STOP);
gst_rtspsrc_loop_send_cmd (src, CMD_STOP, TRUE);
/* stop task if any */
if (src->task) {
gst_task_stop (src->task);
/* make sure it is not running */
g_static_rec_mutex_lock (src->stream_rec_lock);
g_static_rec_mutex_unlock (src->stream_rec_lock);
GST_RTSP_STREAM_LOCK (src);
GST_RTSP_STREAM_UNLOCK (src);
/* no wait for the task to finish */
gst_task_join (src->task);
@ -2925,6 +2956,9 @@ gst_rtspsrc_close (GstRTSPSrc * src)
src->task = NULL;
}
GST_DEBUG_OBJECT (src, "stop flush");
rtsp_connection_flush (src->connection, FALSE);
if (src->methods & RTSP_PLAY) {
/* do TEARDOWN */
res =
@ -3096,7 +3130,6 @@ gst_rtspsrc_play (GstRTSPSrc * src)
* Play Time) and should be put in the NEWSEGMENT position field. */
rtsp_message_get_header (&response, RTSP_HDR_RANGE, &range);
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* FIXME, this is info for the RTP session manager ideally. */
@ -3111,11 +3144,11 @@ gst_rtspsrc_play (GstRTSPSrc * src)
* For UDP we start the task as well to look for server info and UDP timeouts. */
if (src->task == NULL) {
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
gst_task_set_lock (src->task, src->stream_rec_lock);
gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
}
src->running = TRUE;
src->state = RTSP_STATE_PLAYING;
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE);
gst_task_start (src->task);
done:
@ -3227,8 +3260,21 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
const GstStructure *s = gst_message_get_structure (message);
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
gboolean ignore_timeout;
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
GST_OBJECT_LOCK (rtspsrc);
ignore_timeout = rtspsrc->ignore_timeout;
rtspsrc->ignore_timeout = TRUE;
GST_OBJECT_UNLOCK (rtspsrc);
/* we only act on the first udp timeout message, others are irrelevant
* and can be ignored. */
if (ignore_timeout)
gst_message_unref (message);
else
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, TRUE);
return;
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
@ -3300,10 +3346,13 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtspsrc->cur_protocols = rtspsrc->protocols;
/* first attempt, don't ignore timeouts */
rtspsrc->ignore_timeout = FALSE;
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (rtspsrc, "stop flush");
rtsp_connection_flush (rtspsrc->connection, FALSE);
/* FIXME, the server might send UDP packets before we activate the UDP
* ports */
@ -3311,6 +3360,7 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
break;
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);
break;
default:

17
gst/rtsp/gstrtspsrc.h
View file

@ -67,9 +67,13 @@ G_BEGIN_DECLS
typedef struct _GstRTSPSrc GstRTSPSrc;
typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
#define GST_RTSP_STATE_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->state_lock)
#define GST_RTSP_STATE_LOCK(rtsp) (g_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
#define GST_RTSP_STATE_UNLOCK(rtsp) (g_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
#define GST_RTSP_STATE_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->state_rec_lock)
#define GST_RTSP_STATE_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
#define GST_RTSP_STATE_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
#define GST_RTSP_STREAM_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->stream_rec_lock)
#define GST_RTSP_STREAM_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
#define GST_RTSP_STREAM_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
typedef struct _GstRTSPStream GstRTSPStream;
@ -121,9 +125,12 @@ struct _GstRTSPSrc {
gboolean running;
gint free_channel;
/* cond to signal loop */
/* UDP mode loop */
gint loop_cmd;
GMutex *state_lock;
gboolean ignore_timeout;
/* mutex for protecting state changes */
GStaticRecMutex *state_rec_lock;
gint numstreams;
GList *streams;

6
gst/rtsp/rtspconnection.c
View file

@ -205,6 +205,9 @@ rtsp_connection_connect (RTSPConnection * conn, GTimeVal * timeout)
if (fd == -1)
goto sys_error;
/* set to non-blocking mode so that we can cancel the connect */
//fcntl (fd, F_SETFL, O_NONBLOCK);
ret = connect (fd, (struct sockaddr *) &sin, sizeof (sin));
if (ret != 0)
goto sys_error;
@ -216,6 +219,8 @@ rtsp_connection_connect (RTSPConnection * conn, GTimeVal * timeout)
sys_error:
{
if (fd != -1)
CLOSE_SOCKET (fd);
return RTSP_ESYS;
}
not_resolved:
@ -828,7 +833,6 @@ rtsp_connection_close (RTSPConnection * conn)
gint res;
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
g_return_val_if_fail (conn->fd >= 0, RTSP_EINVAL);
if (conn->fd != -1) {
res = CLOSE_SOCKET (conn->fd);