Commit 0dcafb06 authored by Wim Taymans's avatar Wim Taymans
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gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids...

gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455...

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes #455808.
parent 4d581cb6
2007-08-17 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes #455808.
2007-08-17 Wim Taymans <wim.taymans@gmail.com>
* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
......
......@@ -312,9 +312,6 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
static void
gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
{
src->stream_rec_lock = g_new (GStaticRecMutex, 1);
g_static_rec_mutex_init (src->stream_rec_lock);
src->location = g_strdup (DEFAULT_LOCATION);
src->url = NULL;
......@@ -325,8 +322,19 @@ gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
gst_rtsp_ext_list_connect (src->extensions, "send",
(GCallback) gst_rtspsrc_send_cb, src);
/* protects the streaming thread in interleaved mode or the polling
* thread in UDP mode. */
src->stream_rec_lock = g_new (GStaticRecMutex, 1);
g_static_rec_mutex_init (src->stream_rec_lock);
/* protects our state changes from multiple invocations */
src->state_rec_lock = g_new (GStaticRecMutex, 1);
g_static_rec_mutex_init (src->state_rec_lock);
/* protects access to the server connection */
src->conn_rec_lock = g_new (GStaticRecMutex, 1);
g_static_rec_mutex_init (src->conn_rec_lock);
src->state = GST_RTSP_STATE_INVALID;
}
......@@ -338,14 +346,18 @@ gst_rtspsrc_finalize (GObject * object)
rtspsrc = GST_RTSPSRC (object);
gst_rtsp_ext_list_free (rtspsrc->extensions);
g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
g_free (rtspsrc->stream_rec_lock);
g_free (rtspsrc->location);
g_free (rtspsrc->req_location);
g_free (rtspsrc->content_base);
gst_rtsp_url_free (rtspsrc->url);
/* free locks */
g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
g_free (rtspsrc->stream_rec_lock);
g_static_rec_mutex_free (rtspsrc->state_rec_lock);
g_free (rtspsrc->state_rec_lock);
g_static_rec_mutex_free (rtspsrc->conn_rec_lock);
g_free (rtspsrc->conn_rec_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
......@@ -1091,6 +1103,32 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
gst_rtspsrc_push_event (src, event);
}
static GstRTSPResult
gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPMessage * message,
GTimeVal * timeout)
{
GstRTSPResult ret;
GST_RTSP_CONN_LOCK (src);
ret = gst_rtsp_connection_send (src->connection, message, timeout);
GST_RTSP_CONN_UNLOCK (src);
return ret;
}
static GstRTSPResult
gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPMessage * message,
GTimeVal * timeout)
{
GstRTSPResult ret;
GST_RTSP_CONN_LOCK (src);
ret = gst_rtsp_connection_receive (src->connection, message, timeout);
GST_RTSP_CONN_UNLOCK (src);
return ret;
}
static gboolean
gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
{
......@@ -1353,7 +1391,7 @@ gst_rtspsrc_sink_chain (GstPad * pad, GstBuffer * buffer)
gst_rtsp_message_take_body (&message, data, size);
GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
ret = gst_rtsp_connection_send (src->connection, &message, NULL);
ret = gst_rtspsrc_connection_send (src, &message, NULL);
GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
/* and steal it away again because we will free it when unreffing the
......@@ -2218,7 +2256,8 @@ gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPMessage * request)
if (src->debug)
gst_rtsp_message_dump (&response);
if ((res = gst_rtsp_connection_send (src->connection, &response, NULL)) < 0)
res = gst_rtspsrc_connection_send (src, &response, NULL);
if (res < 0)
goto send_error;
return GST_RTSP_OK;
......@@ -2253,7 +2292,8 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
if (src->debug)
gst_rtsp_message_dump (&request);
if ((res = gst_rtsp_connection_send (src->connection, &request, NULL)) < 0)
res = gst_rtspsrc_connection_send (src, &request, NULL);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (src->connection);
......@@ -2305,9 +2345,9 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
GST_DEBUG_OBJECT (src, "doing receive");
res =
gst_rtsp_connection_receive (src->connection, &message,
src->ptcp_timeout);
/* protect the connection with the connection lock so that we can see when
* we are finished doing server communication */
res = gst_rtspsrc_connection_receive (src, &message, src->ptcp_timeout);
switch (res) {
case GST_RTSP_OK:
......@@ -2544,8 +2584,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
/* we should continue reading the TCP socket because the server might
* send us requests. When the session timeout expires, we need to send a
* keep-alive request to keep the session open. */
res =
gst_rtsp_connection_receive (src->connection, &message, &tv_timeout);
res = gst_rtspsrc_connection_receive (src, &message, &tv_timeout);
switch (res) {
case GST_RTSP_OK:
......@@ -2923,17 +2962,15 @@ again:
if (src->debug)
gst_rtsp_message_dump (request);
if ((res =
gst_rtsp_connection_send (src->connection, request,
src->ptcp_timeout)) < 0)
res = gst_rtspsrc_connection_send (src, request, src->ptcp_timeout);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (src->connection);
next:
if ((res =
gst_rtsp_connection_receive (src->connection, response,
src->ptcp_timeout)) < 0)
res = gst_rtspsrc_connection_receive (src, response, src->ptcp_timeout);
if (res < 0)
goto receive_error;
if (src->debug)
......@@ -4099,11 +4136,12 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
if (src->state == GST_RTSP_STATE_READY)
goto was_paused;
/* wait for streaming to finish */
GST_DEBUG_OBJECT (src, "waiting for streaming to finish");
GST_RTSP_STREAM_LOCK (src);
GST_DEBUG_OBJECT (src, "streaming finished");
GST_RTSP_STREAM_UNLOCK (src);
/* waiting for connection idle, we were flushing so any attempt at doing data
* transfer will result in pausing the tasks. */
GST_DEBUG_OBJECT (src, "wait for connection idle");
GST_RTSP_CONN_LOCK (src);
GST_DEBUG_OBJECT (src, "connection is idle now");
GST_RTSP_CONN_UNLOCK (src);
GST_DEBUG_OBJECT (src, "stop connection flush");
gst_rtsp_connection_flush (src->connection, FALSE);
......
......@@ -74,9 +74,13 @@ typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
#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)))
#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)))
#define GST_RTSP_CONN_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->conn_rec_lock)
#define GST_RTSP_CONN_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_CONN_GET_LOCK(rtsp)))
#define GST_RTSP_CONN_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_CONN_GET_LOCK(rtsp)))
typedef struct _GstRTSPStream GstRTSPStream;
......@@ -138,6 +142,9 @@ struct _GstRTSPSrc {
/* mutex for protecting state changes */
GStaticRecMutex *state_rec_lock;
/* mutex for protecting the connection */
GStaticRecMutex *conn_rec_lock;
gint numstreams;
GList *streams;
GstStructure *props;
......
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