rtsp-client: Lock shared media

For shared media we got race conditions. Concurrently rtsp clients might
suspend or unsuspend the shared media and thus change the state without
the clients expecting that.
By introducing a lock that can be taken by callers such as rtsp_client
one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
to handle the media sequentially thus allowing one client to finish its
rtsp call before another client calls on the same media.

https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
Fixes #86

gst/rtsp-server/rtsp-client.c

@@ -1356,6 +1356,7 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
   GstRTSPClientClass *klass;
   GstRTSPSession *session;
   GstRTSPSessionMedia *sessmedia;
+  GstRTSPMedia *media;
   GstRTSPStatusCode code;
   gchar *path;
   gint matched;
@@ -1386,6 +1387,10 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
 
   ctx->sessmedia = sessmedia;
 
+  media = gst_rtsp_session_media_get_media (sessmedia);
+  g_object_ref (media);
+  gst_rtsp_media_lock (media);
+
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST],
       0, ctx, &sig_result);
   if (sig_result != GST_RTSP_STS_OK) {
@@ -1414,6 +1419,9 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
     gst_rtsp_session_pool_remove (priv->session_pool, session);
   }
 
+  gst_rtsp_media_unlock (media);
+  g_object_unref (media);
+
   return TRUE;
 
   /* ERRORS */
@@ -1449,6 +1457,8 @@ sig_failed:
     GST_ERROR ("client %p: pre signal returned error: %s", client,
         gst_rtsp_status_as_text (sig_result));
     send_generic_response (client, sig_result, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 }
@@ -1617,6 +1627,8 @@ handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
   g_free (path);
 
   media = gst_rtsp_session_media_get_media (sessmedia);
+  g_object_ref (media);
+  gst_rtsp_media_lock (media);
   n = gst_rtsp_media_n_streams (media);
   for (i = 0; i < n; i++) {
     GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
@@ -1655,6 +1667,9 @@ handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
 
+  gst_rtsp_media_unlock (media);
+  g_object_unref (media);
+
   return TRUE;
 
   /* ERRORS */
@@ -1690,6 +1705,8 @@ sig_failed:
     GST_ERROR ("client %p: pre signal returned error: %s", client,
         gst_rtsp_status_as_text (sig_result));
     send_generic_response (client, sig_result, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 invalid_state:
@@ -1697,12 +1714,16 @@ invalid_state:
     GST_ERROR ("client %p: not PLAYING or RECORDING", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
         ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 not_supported:
   {
     GST_ERROR ("client %p: pausing not supported", client);
     send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 }
@@ -1974,6 +1995,9 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
   ctx->sessmedia = sessmedia;
   ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
 
+  g_object_ref (media);
+  gst_rtsp_media_lock (media);
+
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST], 0,
       ctx, &sig_result);
   if (sig_result != GST_RTSP_STS_OK) {
@@ -2060,6 +2084,9 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
 
+  gst_rtsp_media_unlock (media);
+  g_object_unref (media);
+
   return TRUE;
 
   /* ERRORS */
@@ -2094,6 +2121,8 @@ sig_failed:
     GST_ERROR ("client %p: pre signal returned error: %s", client,
         gst_rtsp_status_as_text (sig_result));
     send_generic_response (client, sig_result, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 invalid_state:
@@ -2101,6 +2130,8 @@ invalid_state:
     GST_ERROR ("client %p: not PLAYING or READY", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
         ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 pipeline_error:
@@ -2108,42 +2139,56 @@ pipeline_error:
     GST_ERROR ("client %p: failed to configure the pipeline", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
         ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 unsuspend_failed:
   {
     GST_ERROR ("client %p: unsuspend failed", client);
     send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 invalid_mode:
   {
     GST_ERROR ("client %p: seek failed", client);
     send_generic_response (client, code, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 unsupported_mode:
   {
     GST_ERROR ("client %p: media does not support PLAY", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 get_rates_error:
   {
     GST_ERROR ("client %p: failed obtaining rate and applied_rate", client);
     send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 adjust_play_response_failed:
   {
     GST_ERROR ("client %p: failed to adjust play response", client);
     send_generic_response (client, code, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 rtp_info_error:
   {
     GST_ERROR ("client %p: failed to add RTP-Info", client);
     send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 }
@@ -2648,13 +2693,17 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
     /* get a handle to the configuration of the media in the session */
     media = find_media (client, ctx, path, &matched);
     /* need to suspend the media, if the protocol has changed */
-    if (media != NULL)
+    if (media != NULL) {
+      gst_rtsp_media_lock (media);
       gst_rtsp_media_suspend (media);
+    }
   } else {
-    if ((media = gst_rtsp_session_media_get_media (sessmedia)))
+    if ((media = gst_rtsp_session_media_get_media (sessmedia))) {
       g_object_ref (media);
-    else
+      gst_rtsp_media_lock (media);
+    } else {
       goto media_not_found;
+    }
   }
   /* no media, not found then */
   if (media == NULL)
@@ -2870,12 +2919,14 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
       gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
       break;
   }
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
+
+  gst_rtsp_media_unlock (media);
   g_object_unref (media);
   g_object_unref (session);
   g_free (path);
 
-  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
-
   return TRUE;
 
   /* ERRORS */
@@ -2913,6 +2964,7 @@ control_not_found:
   {
     GST_ERROR ("client %p: no control in path '%s'", client, path);
     send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     goto cleanup_session;
   }
@@ -2921,6 +2973,7 @@ stream_not_found:
     GST_ERROR ("client %p: stream '%s' not found", client,
         GST_STR_NULL (control));
     send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     goto cleanup_session;
   }
@@ -2929,6 +2982,7 @@ sig_failed:
     GST_ERROR ("client %p: pre signal returned error: %s", client,
         gst_rtsp_status_as_text (sig_result));
     send_generic_response (client, sig_result, ctx);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     goto cleanup_path;
   }
@@ -2936,6 +2990,7 @@ service_unavailable:
   {
     GST_ERROR ("client %p: can't create session", client);
     send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     goto cleanup_session;
   }
@@ -2948,6 +3003,7 @@ sessmedia_unavailable:
 configure_media_failed_no_reply:
   {
     GST_ERROR ("client %p: configure_media failed", client);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     /* error reply is already sent */
     goto cleanup_session;
@@ -2991,8 +3047,10 @@ keymgmt_error:
   {
   cleanup_transport:
     gst_rtsp_transport_free (ct);
-    if (media)
+    if (media) {
+      gst_rtsp_media_unlock (media);
       g_object_unref (media);
+    }
   cleanup_session:
     if (new_session)
       gst_rtsp_session_pool_remove (priv->session_pool, session);
@@ -3105,6 +3163,8 @@ handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
   if (!(media = find_media (client, ctx, path, NULL)))
     goto no_media;
 
+  gst_rtsp_media_lock (media);
+
   if (!(gst_rtsp_media_get_transport_mode (media) &
           GST_RTSP_TRANSPORT_MODE_PLAY))
     goto unsupported_mode;
@@ -3115,7 +3175,6 @@ handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
 
   /* we suspend after the describe */
   gst_rtsp_media_suspend (media);
-  g_object_unref (media);
 
   gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
       gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
@@ -3140,6 +3199,9 @@ handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
       0, ctx);
 
+  gst_rtsp_media_unlock (media);
+  g_object_unref (media);
+
   return TRUE;
 
   /* ERRORS */
@@ -3180,6 +3242,7 @@ unsupported_mode:
     GST_ERROR ("client %p: media does not support DESCRIBE", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
     g_free (path);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     return FALSE;
   }
@@ -3188,6 +3251,7 @@ no_sdp:
     GST_ERROR ("client %p: can't create SDP", client);
     send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
     g_free (path);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     return FALSE;
   }
@@ -3284,6 +3348,7 @@ handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
     goto no_media;
 
   ctx->media = media;
+  gst_rtsp_media_lock (media);
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST],
       0, ctx, &sig_result);
@@ -3319,7 +3384,6 @@ handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
 
   /* we suspend after the announce */
   gst_rtsp_media_suspend (media);
-  g_object_unref (media);
 
   send_message (client, ctx, ctx->response, FALSE);
 
@@ -3328,6 +3392,9 @@ handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
 
   gst_sdp_message_free (sdp);
   g_free (path);
+  gst_rtsp_media_unlock (media);
+  g_object_unref (media);
+
   return TRUE;
 
 no_uri:
@@ -3382,6 +3449,8 @@ sig_failed:
         gst_rtsp_status_as_text (sig_result));
     send_generic_response (client, sig_result, ctx);
     gst_sdp_message_free (sdp);
+    gst_rtsp_media_unlock (media);
+    g_object_unref (media);
     return FALSE;
   }
 unsupported_mode:
@@ -3389,6 +3458,7 @@ unsupported_mode:
     GST_ERROR ("client %p: media does not support ANNOUNCE", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
     g_free (path);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     gst_sdp_message_free (sdp);
     return FALSE;
@@ -3398,6 +3468,7 @@ unhandled_sdp:
     GST_ERROR ("client %p: can't handle SDP", client);
     send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
     g_free (path);
+    gst_rtsp_media_unlock (media);
     g_object_unref (media);
     gst_sdp_message_free (sdp);
     return FALSE;

gst/rtsp-server/rtsp-media.c

@@ -90,6 +90,12 @@ struct _GstRTSPMediaPrivate
   GMutex lock;
   GCond cond;
 
+  /* the global lock is used to lock the entire media. This is needed by callers
+     such as rtsp_client to protect the media when it is shared by many clients.
+     The lock prevents that concurrenting clients messes up media.
+     Typically the lock is taken in external API calls such as SETUP */
+  GMutex global_lock;
+
   /* protected by lock */
   GstRTSPPermissions *permissions;
   gboolean shared;
@@ -459,6 +465,7 @@ gst_rtsp_media_init (GstRTSPMedia * media)
 
   priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
   g_mutex_init (&priv->lock);
+  g_mutex_init (&priv->global_lock);
   g_cond_init (&priv->cond);
   g_rec_mutex_init (&priv->state_lock);
 
@@ -512,6 +519,7 @@ gst_rtsp_media_finalize (GObject * obj)
     gst_object_unref (priv->clock);
   g_free (priv->multicast_iface);
   g_mutex_clear (&priv->lock);
+  g_mutex_clear (&priv->global_lock);
   g_cond_clear (&priv->cond);
   g_rec_mutex_clear (&priv->state_lock);
 
@@ -3994,6 +4002,54 @@ get_clock_unlocked (GstRTSPMedia * media)
   return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
 }
 
+/**
+ * gst_rtsp_media_lock:
+ * @media: a #GstRTSPMedia
+ *
+ * Lock the entire media. This is needed by callers such as rtsp_client to
+ * protect the media when it is shared by many clients.
+ * The lock prevents that concurrent clients alters the shared media,
+ * while one client already is working with it.
+ * Typically the lock is taken in external RTSP API calls that uses shared media
+ * such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE.
+ *
+ * As best practice take the lock as soon as the function get hold of a shared
+ * media object. Release the lock right before the function returns.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_lock (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->global_lock);
+}
+
+/**
+ * gst_rtsp_media_unlock:
+ * @media: a #GstRTSPMedia
+ *
+ * Unlock the media.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_unlock (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_unlock (&priv->global_lock);
+}
+
 /**
  * gst_rtsp_media_get_clock:
  * @media: a #GstRTSPMedia
@@ -4369,6 +4425,14 @@ gst_rtsp_media_suspend (GstRTSPMedia * media)
 
   GST_FIXME ("suspend for dynamic pipelines needs fixing");
 
+  /* this typically can happen for shared media. */
+  if (priv->prepare_count > 1 &&
+      priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED) {
+    goto done;
+  } else if (priv->prepare_count > 1) {
+    goto prepared_by_other_client;
+  }
+
   g_rec_mutex_lock (&priv->state_lock);
   if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
     goto not_prepared;
@@ -4390,6 +4454,11 @@ done:
   return TRUE;
 
   /* ERRORS */
+prepared_by_other_client:
+  {
+    GST_WARNING ("media %p was prepared by other client", media);
+    return FALSE;
+  }
 not_prepared:
   {
     g_rec_mutex_unlock (&priv->state_lock);

gst/rtsp-server/rtsp-media.h

@@ -364,6 +364,12 @@ GstRTSPStream *       gst_rtsp_media_create_stream    (GstRTSPMedia *media,
 
 /* dealing with the media */
 
+GST_RTSP_SERVER_API
+void                  gst_rtsp_media_lock             (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void                  gst_rtsp_media_unlock           (GstRTSPMedia *media);
+
 GST_RTSP_SERVER_API
 GstClock *            gst_rtsp_media_get_clock        (GstRTSPMedia *media);