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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.
This commit is contained in:
Wim Taymans
parent
4d581cb606
commit
0dcafb0635
3 changed files with 84 additions and 27 deletions
12
ChangeLog
12
ChangeLog
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@ -1,3 +1,15 @@
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2007-08-17 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
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(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
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(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
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(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
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(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
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(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
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* gst/rtsp/gstrtspsrc.h:
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Protect connection activity with a new lock, avoids deadlocks when going
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to PAUSED. Fixes #455808.
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2007-08-17 Wim Taymans <wim.taymans@gmail.com>
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* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
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@ -312,9 +312,6 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
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static void
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gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
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{
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src->stream_rec_lock = g_new (GStaticRecMutex, 1);
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g_static_rec_mutex_init (src->stream_rec_lock);
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src->location = g_strdup (DEFAULT_LOCATION);
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src->url = NULL;
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@ -325,8 +322,19 @@ gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
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gst_rtsp_ext_list_connect (src->extensions, "send",
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(GCallback) gst_rtspsrc_send_cb, src);
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/* protects the streaming thread in interleaved mode or the polling
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* thread in UDP mode. */
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src->stream_rec_lock = g_new (GStaticRecMutex, 1);
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g_static_rec_mutex_init (src->stream_rec_lock);
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/* protects our state changes from multiple invocations */
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src->state_rec_lock = g_new (GStaticRecMutex, 1);
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g_static_rec_mutex_init (src->state_rec_lock);
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/* protects access to the server connection */
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src->conn_rec_lock = g_new (GStaticRecMutex, 1);
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g_static_rec_mutex_init (src->conn_rec_lock);
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src->state = GST_RTSP_STATE_INVALID;
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}
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@ -338,14 +346,18 @@ gst_rtspsrc_finalize (GObject * object)
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rtspsrc = GST_RTSPSRC (object);
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gst_rtsp_ext_list_free (rtspsrc->extensions);
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g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
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g_free (rtspsrc->stream_rec_lock);
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g_free (rtspsrc->location);
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g_free (rtspsrc->req_location);
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g_free (rtspsrc->content_base);
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gst_rtsp_url_free (rtspsrc->url);
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/* free locks */
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g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
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g_free (rtspsrc->stream_rec_lock);
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g_static_rec_mutex_free (rtspsrc->state_rec_lock);
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g_free (rtspsrc->state_rec_lock);
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g_static_rec_mutex_free (rtspsrc->conn_rec_lock);
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g_free (rtspsrc->conn_rec_lock);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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@ -1091,6 +1103,32 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
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gst_rtspsrc_push_event (src, event);
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}
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static GstRTSPResult
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gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPMessage * message,
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GTimeVal * timeout)
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{
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GstRTSPResult ret;
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GST_RTSP_CONN_LOCK (src);
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ret = gst_rtsp_connection_send (src->connection, message, timeout);
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GST_RTSP_CONN_UNLOCK (src);
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return ret;
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}
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static GstRTSPResult
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gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPMessage * message,
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GTimeVal * timeout)
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{
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GstRTSPResult ret;
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GST_RTSP_CONN_LOCK (src);
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ret = gst_rtsp_connection_receive (src->connection, message, timeout);
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GST_RTSP_CONN_UNLOCK (src);
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return ret;
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}
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static gboolean
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gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
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{
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@ -1353,7 +1391,7 @@ gst_rtspsrc_sink_chain (GstPad * pad, GstBuffer * buffer)
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gst_rtsp_message_take_body (&message, data, size);
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GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
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ret = gst_rtsp_connection_send (src->connection, &message, NULL);
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ret = gst_rtspsrc_connection_send (src, &message, NULL);
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GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
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/* and steal it away again because we will free it when unreffing the
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@ -2218,7 +2256,8 @@ gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPMessage * request)
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if (src->debug)
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gst_rtsp_message_dump (&response);
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if ((res = gst_rtsp_connection_send (src->connection, &response, NULL)) < 0)
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res = gst_rtspsrc_connection_send (src, &response, NULL);
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if (res < 0)
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goto send_error;
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return GST_RTSP_OK;
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@ -2253,7 +2292,8 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
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if (src->debug)
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gst_rtsp_message_dump (&request);
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if ((res = gst_rtsp_connection_send (src->connection, &request, NULL)) < 0)
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res = gst_rtspsrc_connection_send (src, &request, NULL);
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if (res < 0)
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goto send_error;
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gst_rtsp_connection_reset_timeout (src->connection);
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@ -2305,9 +2345,9 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
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GST_DEBUG_OBJECT (src, "doing receive");
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res =
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gst_rtsp_connection_receive (src->connection, &message,
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src->ptcp_timeout);
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/* protect the connection with the connection lock so that we can see when
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* we are finished doing server communication */
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res = gst_rtspsrc_connection_receive (src, &message, src->ptcp_timeout);
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switch (res) {
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case GST_RTSP_OK:
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@ -2544,8 +2584,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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/* we should continue reading the TCP socket because the server might
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* send us requests. When the session timeout expires, we need to send a
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* keep-alive request to keep the session open. */
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res =
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gst_rtsp_connection_receive (src->connection, &message, &tv_timeout);
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res = gst_rtspsrc_connection_receive (src, &message, &tv_timeout);
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switch (res) {
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case GST_RTSP_OK:
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@ -2923,17 +2962,15 @@ again:
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if (src->debug)
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gst_rtsp_message_dump (request);
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if ((res =
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gst_rtsp_connection_send (src->connection, request,
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src->ptcp_timeout)) < 0)
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res = gst_rtspsrc_connection_send (src, request, src->ptcp_timeout);
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if (res < 0)
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goto send_error;
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gst_rtsp_connection_reset_timeout (src->connection);
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next:
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if ((res =
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gst_rtsp_connection_receive (src->connection, response,
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src->ptcp_timeout)) < 0)
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res = gst_rtspsrc_connection_receive (src, response, src->ptcp_timeout);
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if (res < 0)
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goto receive_error;
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if (src->debug)
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@ -4099,11 +4136,12 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
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if (src->state == GST_RTSP_STATE_READY)
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goto was_paused;
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/* wait for streaming to finish */
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GST_DEBUG_OBJECT (src, "waiting for streaming to finish");
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GST_RTSP_STREAM_LOCK (src);
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GST_DEBUG_OBJECT (src, "streaming finished");
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GST_RTSP_STREAM_UNLOCK (src);
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/* waiting for connection idle, we were flushing so any attempt at doing data
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* transfer will result in pausing the tasks. */
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GST_DEBUG_OBJECT (src, "wait for connection idle");
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GST_RTSP_CONN_LOCK (src);
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GST_DEBUG_OBJECT (src, "connection is idle now");
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GST_RTSP_CONN_UNLOCK (src);
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GST_DEBUG_OBJECT (src, "stop connection flush");
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gst_rtsp_connection_flush (src->connection, FALSE);
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@ -78,6 +78,10 @@ typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
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#define GST_RTSP_STREAM_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
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#define GST_RTSP_STREAM_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
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#define GST_RTSP_CONN_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->conn_rec_lock)
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#define GST_RTSP_CONN_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_CONN_GET_LOCK(rtsp)))
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#define GST_RTSP_CONN_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_CONN_GET_LOCK(rtsp)))
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typedef struct _GstRTSPStream GstRTSPStream;
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struct _GstRTSPStream {
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@ -138,6 +142,9 @@ struct _GstRTSPSrc {
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/* mutex for protecting state changes */
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GStaticRecMutex *state_rec_lock;
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/* mutex for protecting the connection */
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GStaticRecMutex *conn_rec_lock;
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gint numstreams;
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GList *streams;
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GstStructure *props;
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