gst/rtsp/gstrtspsrc.*: Protect state changes with a lock.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect state changes with a lock.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(parse_line):
* gst/rtsp/rtspconnection.h:
Remove some unused stuff.

5 files changed, 102 insertions, 8 deletions

diff --git a/ChangeLog b/ChangeLog
index 1b1a2cdd..98d487b7 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,20 @@
 2007-04-26  Wim Taymans  <wim@fluendo.com>
 
+	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
+	(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
+	(gst_rtspsrc_open), (gst_rtspsrc_close),
+	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
+	(gst_rtspsrc_pause):
+	* gst/rtsp/gstrtspsrc.h:
+	Protect state changes with a lock.
+
+	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
+	(parse_line):
+	* gst/rtsp/rtspconnection.h:
+	Remove some unused stuff.
+
+2007-04-26  Wim Taymans  <wim@fluendo.com>
+
 	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
 	Handle the case where there are exactly 0 bytes to read and the ioctl
 	did not report an error. Fixes #433530.
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index ef79b6e3..85966dbd 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -302,6 +302,9 @@ gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
   src->extension = rtsp_ext_wms_get_context ();
 #endif
   src->extension->src = (gpointer) src;
+
+  src->state_lock = g_mutex_new ();
+  src->state = RTSP_STATE_INVALID;
 }
 
 static void
@@ -319,6 +322,7 @@ 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);
 
   if (rtspsrc->extension) {
 #ifdef WITH_EXT_REAL
@@ -2554,6 +2558,8 @@ gst_rtspsrc_open (GstRTSPSrc * src)
   GstRTSPStream *stream = NULL;
   gchar *respcont = NULL;
 
+  GST_RTSP_STATE_LOCK (src);
+
   /* reset our state */
   gst_segment_init (&src->segment, GST_FORMAT_TIME);
 
@@ -2645,8 +2651,14 @@ gst_rtspsrc_open (GstRTSPSrc * src)
     stream = gst_rtspsrc_create_stream (src, &sdp, i);
   }
 
+  src->state = RTSP_STATE_INIT;
+
   /* setup streams */
-  gst_rtspsrc_setup_streams (src);
+  if (!gst_rtspsrc_setup_streams (src))
+    goto setup_failed;
+
+  src->state = RTSP_STATE_READY;
+  GST_RTSP_STATE_UNLOCK (src);
 
   /* clean up any messages */
   rtsp_message_unset (&request);
@@ -2705,8 +2717,14 @@ wrong_content_type:
         ("Server does not support SDP, got %s.", respcont));
     goto cleanup_error;
   }
+setup_failed:
+  {
+    /* error was posted */
+    goto cleanup_error;
+  }
 cleanup_error:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     rtsp_message_unset (&request);
     rtsp_message_unset (&response);
     return FALSE;
@@ -2722,6 +2740,8 @@ gst_rtspsrc_close (GstRTSPSrc * src)
 
   GST_DEBUG_OBJECT (src, "TEARDOWN...");
 
+  GST_RTSP_STATE_LOCK (src);
+
   gst_rtspsrc_loop_send_cmd (src, CMD_STOP);
 
   /* stop task if any */
@@ -2767,17 +2787,22 @@ gst_rtspsrc_close (GstRTSPSrc * src)
   /* cleanup */
   gst_rtspsrc_cleanup (src);
 
+  src->state = RTSP_STATE_INVALID;
+  GST_RTSP_STATE_UNLOCK (src);
+
   return TRUE;
 
   /* ERRORS */
 create_request_failed:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request."));
     return FALSE;
   }
 send_error:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     rtsp_message_unset (&request);
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
         ("Could not send message."));
@@ -2785,6 +2810,7 @@ send_error:
   }
 close_failed:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, (NULL), ("Close failed."));
     return FALSE;
   }
@@ -2874,11 +2900,16 @@ gst_rtspsrc_play (GstRTSPSrc * src)
   RTSPResult res;
   gchar *rtpinfo;
 
-  if (!(src->methods & RTSP_PLAY))
-    return TRUE;
+  GST_RTSP_STATE_LOCK (src);
 
   GST_DEBUG_OBJECT (src, "PLAY...");
 
+  if (!(src->methods & RTSP_PLAY))
+    goto not_supported;
+
+  if (src->state == RTSP_STATE_PLAYING)
+    goto was_playing;
+
   /* do play */
   res = rtsp_message_init_request (&request, RTSP_PLAY, src->req_location);
   if (res < 0)
@@ -2908,20 +2939,36 @@ gst_rtspsrc_play (GstRTSPSrc * src)
     gst_task_set_lock (src->task, src->stream_rec_lock);
   }
   src->running = TRUE;
+  src->state = RTSP_STATE_PLAYING;
   gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
   gst_task_start (src->task);
 
+done:
+  GST_RTSP_STATE_UNLOCK (src);
+
   return TRUE;
 
   /* ERRORS */
+not_supported:
+  {
+    GST_DEBUG_OBJECT (src, "PLAY is not supported");
+    goto done;
+  }
+was_playing:
+  {
+    GST_DEBUG_OBJECT (src, "we were already PLAYING");
+    goto done;
+  }
 create_request_failed:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request."));
     return FALSE;
   }
 send_error:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     rtsp_message_unset (&request);
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
         ("Could not send message."));
@@ -2936,10 +2983,16 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
   RTSPMessage response = { 0 };
   RTSPResult res;
 
-  if (!(src->methods & RTSP_PAUSE))
-    return TRUE;
+  GST_RTSP_STATE_LOCK (src);
 
   GST_DEBUG_OBJECT (src, "PAUSE...");
+
+  if (!(src->methods & RTSP_PAUSE))
+    goto not_supported;
+
+  if (src->state == RTSP_STATE_READY)
+    goto was_paused;
+
   /* do pause */
   res = rtsp_message_init_request (&request, RTSP_PAUSE, src->req_location);
   if (res < 0)
@@ -2951,17 +3004,34 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
   rtsp_message_unset (&request);
   rtsp_message_unset (&response);
 
+  src->state = RTSP_STATE_READY;
+
+done:
+  GST_RTSP_STATE_UNLOCK (src);
+
   return TRUE;
 
   /* ERRORS */
+not_supported:
+  {
+    GST_DEBUG_OBJECT (src, "PAUSE is not supported");
+    goto done;
+  }
+was_paused:
+  {
+    GST_DEBUG_OBJECT (src, "we were already PAUSED");
+    goto done;
+  }
 create_request_failed:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request."));
     return FALSE;
   }
 send_error:
   {
+    GST_RTSP_STATE_UNLOCK (src);
     rtsp_message_unset (&request);
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
         ("Could not send message."));
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 82582cf3..7d595f69 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -67,6 +67,10 @@ 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_LOOP_GET_COND(rtsp)     (GST_RTSPSRC_CAST(rtsp)->loop_cond)
 #define GST_RTSP_LOOP_WAIT(rtsp)         (g_cond_wait(GST_RTSP_LOOP_GET_COND (rtsp), GST_OBJECT_GET_LOCK (rtsp)))
 #define GST_RTSP_LOOP_SIGNAL(rtsp)       (g_cond_signal(GST_RTSP_LOOP_GET_COND (rtsp)))
@@ -124,6 +128,7 @@ struct _GstRTSPSrc {
   /* cond to signal loop */
   GCond           *loop_cond;
   gint             loop_cmd;
+  GMutex          *state_lock;
 
   gint             numstreams;
   GList           *streams;
@@ -141,6 +146,7 @@ struct _GstRTSPSrc {
   guint            latency;
 
   /* state */
+  RTSPState        state;
   gchar           *content_base;
   RTSPLowerTrans   cur_protocols;
   gboolean         tried_url_auth;
diff --git a/gst/rtsp/rtspconnection.c b/gst/rtsp/rtspconnection.c
index e921a346..448dccd3 100644
--- a/gst/rtsp/rtspconnection.c
+++ b/gst/rtsp/rtspconnection.c
@@ -142,7 +142,6 @@ rtsp_connection_create (RTSPUrl * url, RTSPConnection ** conn)
   newconn->fd = -1;
   newconn->cseq = 0;
   newconn->session_id[0] = 0;
-  newconn->state = RTSP_STATE_INIT;
 
   newconn->auth_method = RTSP_AUTH_NONE;
   newconn->username = NULL;
@@ -524,7 +523,7 @@ parse_line (gchar * buffer, RTSPMessage * msg)
   /* read key */
   read_key (key, sizeof (key), &bptr);
   if (*bptr != ':')
-    return RTSP_EINVAL;
+    goto no_column;
 
   bptr++;
 
@@ -536,6 +535,11 @@ parse_line (gchar * buffer, RTSPMessage * msg)
   }
 
   return RTSP_OK;
+
+no_column:
+  {
+    return RTSP_EINVAL;
+  }
 }
 
 RTSPResult
diff --git a/gst/rtsp/rtspconnection.h b/gst/rtsp/rtspconnection.h
index 2076950b..fc3b28cf 100644
--- a/gst/rtsp/rtspconnection.h
+++ b/gst/rtsp/rtspconnection.h
@@ -61,7 +61,6 @@ typedef struct _RTSPConnection
   gint control_sock[2];
 
   /* Session state */
-  RTSPState     state;
   gint          cseq;                   /* sequence number */
   gchar         session_id[512];        /* session id */