FaSTTUBe/GST-Tensordecoder-ov_ep
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webrtc: Fix a race on shutdown.

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The main context can disappear in gst_webrtc_bin_enqueue_task()
between checking the is_closed flag and enqueueing a source on the
main context. Protect the main context with the object lock instead
of the PC lock, and hold a ref briefly to make sure it stays alive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
This commit is contained in:
Jan Schmidt 2020-10-30 16:31:18 +11:00 committed by GStreamer Merge Bot
parent 80a56c25a6
commit af90778314
1 changed files with 16 additions and 3 deletions
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19
ext/webrtc/gstwebrtcbin.c
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@ -713,9 +713,12 @@ _gst_pc_thread (GstWebRTCBin * webrtc)
* tasks */ * tasks */
g_main_loop_run (webrtc->priv->loop); g_main_loop_run (webrtc->priv->loop);
PC_LOCK (webrtc); GST_OBJECT_LOCK (webrtc);
g_main_context_unref (webrtc->priv->main_context); g_main_context_unref (webrtc->priv->main_context);
webrtc->priv->main_context = NULL; webrtc->priv->main_context = NULL;
GST_OBJECT_UNLOCK (webrtc);
PC_LOCK (webrtc);
g_main_loop_unref (webrtc->priv->loop); g_main_loop_unref (webrtc->priv->loop);
webrtc->priv->loop = NULL; webrtc->priv->loop = NULL;
PC_COND_BROADCAST (webrtc); PC_COND_BROADCAST (webrtc);
@ -744,8 +747,11 @@ _start_thread (GstWebRTCBin * webrtc)
static void static void
_stop_thread (GstWebRTCBin * webrtc) _stop_thread (GstWebRTCBin * webrtc)
{ {
PC_LOCK (webrtc); GST_OBJECT_LOCK (webrtc);
webrtc->priv->is_closed = TRUE; webrtc->priv->is_closed = TRUE;
GST_OBJECT_UNLOCK (webrtc);
PC_LOCK (webrtc);
g_main_loop_quit (webrtc->priv->loop); g_main_loop_quit (webrtc->priv->loop);
while (webrtc->priv->loop) while (webrtc->priv->loop)
PC_COND_WAIT (webrtc); PC_COND_WAIT (webrtc);
@ -804,16 +810,22 @@ gst_webrtc_bin_enqueue_task (GstWebRTCBin * webrtc, GstWebRTCBinFunc func,
gpointer data, GDestroyNotify notify, GstPromise * promise) gpointer data, GDestroyNotify notify, GstPromise * promise)
{ {
GstWebRTCBinTask *op; GstWebRTCBinTask *op;
GMainContext *ctx;
GSource *source; GSource *source;
g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE); g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE);
GST_OBJECT_LOCK (webrtc);
if (webrtc->priv->is_closed) { if (webrtc->priv->is_closed) {
GST_OBJECT_UNLOCK (webrtc);
GST_DEBUG_OBJECT (webrtc, "Peerconnection is closed, aborting execution"); GST_DEBUG_OBJECT (webrtc, "Peerconnection is closed, aborting execution");
if (notify) if (notify)
notify (data); notify (data);
return FALSE; return FALSE;
} }
ctx = g_main_context_ref (webrtc->priv->main_context);
GST_OBJECT_UNLOCK (webrtc);
op = g_new0 (GstWebRTCBinTask, 1); op = g_new0 (GstWebRTCBinTask, 1);
op->webrtc = webrtc; op->webrtc = webrtc;
op->op = func; op->op = func;
@ -826,8 +838,9 @@ gst_webrtc_bin_enqueue_task (GstWebRTCBin * webrtc, GstWebRTCBinFunc func,
g_source_set_priority (source, G_PRIORITY_DEFAULT); g_source_set_priority (source, G_PRIORITY_DEFAULT);
g_source_set_callback (source, (GSourceFunc) _execute_op, op, g_source_set_callback (source, (GSourceFunc) _execute_op, op,
(GDestroyNotify) _free_op); (GDestroyNotify) _free_op);
g_source_attach (source, webrtc->priv->main_context); g_source_attach (source, ctx);
g_source_unref (source); g_source_unref (source);
g_main_context_unref (ctx);
return TRUE; return TRUE;
} }