rtspsrc: open on play and pause when not done yet

With the async state changes, it is possible that we need to open the stream
before play and pause.
Also make sure we remember a previous open failure so that we don't keep trying
again.

gst/rtsp/gstrtspsrc.c

@@ -3952,10 +3952,6 @@ gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
   /* we can try only TCP now */
   src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
 
-  /* pause to prepare for a restart */
-  if ((res = gst_rtspsrc_pause (src, FALSE, async)) < 0)
-    goto done;
-
   /* close and cleanup our state */
   if ((res = gst_rtspsrc_close (src, async)) < 0)
     goto done;
@@ -5744,11 +5740,13 @@ done:
 no_sdp:
   {
     GST_WARNING_OBJECT (src, "can't get sdp");
+    src->open_error = TRUE;
     goto done;
   }
 open_failed:
   {
     GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
+    src->open_error = TRUE;
     goto done;
   }
 }
@@ -5986,6 +5984,30 @@ clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
   }
 }
 
+static GstRTSPResult
+gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+
+  if (src->state < GST_RTSP_STATE_READY) {
+    res = GST_RTSP_ERROR;
+    if (src->open_error) {
+      GST_DEBUG_OBJECT (src, "the stream was in error");
+      goto done;
+    }
+    if (async)
+      gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
+
+    if ((res = gst_rtspsrc_open (src, async)) < 0) {
+      GST_DEBUG_OBJECT (src, "failed to open stream");
+      goto done;
+    }
+  }
+
+done:
+  return res;
+}
+
 static GstRTSPResult
 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
 {
@@ -5999,6 +6021,9 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
 
   GST_DEBUG_OBJECT (src, "PLAY...");
 
+  if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
+    goto open_failed;
+
   if (!(src->methods & GST_RTSP_PLAY))
     goto not_supported;
 
@@ -6149,6 +6174,11 @@ done:
   return res;
 
   /* ERRORS */
+open_failed:
+  {
+    GST_DEBUG_OBJECT (src, "failed to open stream");
+    goto done;
+  }
 not_supported:
   {
     GST_DEBUG_OBJECT (src, "PLAY is not supported");
@@ -6195,6 +6225,9 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
 
   GST_DEBUG_OBJECT (src, "PAUSE...");
 
+  if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
+    goto open_failed;
+
   if (!(src->methods & GST_RTSP_PAUSE))
     goto not_supported;
 
@@ -6261,6 +6294,11 @@ done:
   return res;
 
   /* ERRORS */
+open_failed:
+  {
+    GST_DEBUG_OBJECT (src, "failed to open stream");
+    goto done;
+  }
 not_supported:
   {
     GST_DEBUG_OBJECT (src, "PAUSE is not supported");
@@ -6395,6 +6433,7 @@ gst_rtspsrc_thread (GstRTSPSrc * src)
       src->cur_protocols = src->protocols;
       /* first attempt, don't ignore timeouts */
       src->ignore_timeout = FALSE;
+      src->open_error = FALSE;
       ret = gst_rtspsrc_open (src, TRUE);
       break;
     case CMD_PLAY:

gst/rtsp/gstrtspsrc.h

@@ -182,6 +182,7 @@ struct _GstRTSPSrc {
   gboolean         ignore_timeout;
   gboolean         flushing;
   gboolean         waiting;
+  gboolean         open_error;
 
   /* mutex for protecting state changes */
   GStaticRecMutex *state_rec_lock;