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webrtcbin: Fix deadlock when receiving new sctp stream

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When receiving an sctp message for a stream that not yet has an
sctpdec pad associated with it means we end up in
_on_sctpdec_pad_added. At this point we're holding the sctpassocation
lock. Then it's not possible to take the pc_lock because then code
executing under the pc_lock (which means anything in the webrtc
thread) may not take the sctpassociation lock. For example, running
the data channel close procedure from the webrtc thread means we
eventually end up sending a SCTP_RESET_STREAMS packet which needs to
grab the sctpassociation lock.

This means _on_sctpdec_pad_added simply cannot take the pc_lock and
also it is not possible to postpone the channel creation as we need to
link the pads right there. The solution is to introduce a more
granular dc_lock that protects only the things that needs to be done
to create the datachannel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186>
This commit is contained in:
Johan Sternerup 2021-04-22 10:43:55 +02:00 committed by GStreamer Marge Bot
parent 8dbdfad914
commit 4d514abfd6
2 changed files with 49 additions and 30 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

76
ext/webrtc/gstwebrtcbin.c
View file

@ -55,6 +55,9 @@
#define ICE_LOCK(w) (g_mutex_lock (ICE_GET_LOCK(w)))
#define ICE_UNLOCK(w) (g_mutex_unlock (ICE_GET_LOCK(w)))
#define DC_GET_LOCK(w) (&w->priv->dc_lock)
#define DC_LOCK(w) (g_mutex_lock (DC_GET_LOCK(w)))
#define DC_UNLOCK(w) (g_mutex_unlock (DC_GET_LOCK(w)))
/* The extra time for the rtpstorage compared to the RTP jitterbuffer (in ms) */
#define RTPSTORAGE_EXTRA_TIME (50)
@ -579,6 +582,7 @@ data_channel_match_for_id (WebRTCDataChannel * channel, gint * id)
return channel->parent.id == *id;
}
/* always called with dc_lock held */
static WebRTCDataChannel *
_find_data_channel_for_id (GstWebRTCBin * webrtc, gint id)
{
@ -1801,12 +1805,14 @@ gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc)
if (!webrtc->priv->sctp_transport)
return;
DC_LOCK (webrtc);
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
GstWebRTCDataChannel *channel
= g_ptr_array_index (webrtc->priv->data_channels, i);
sctp_priority = MAX (sctp_priority, channel->priority);
}
DC_UNLOCK (webrtc);
/* Default priority is low means DSCP field is left as 0 */
if (sctp_priority == 0)
@ -1961,13 +1967,16 @@ _on_data_channel_ready_state (WebRTCDataChannel * channel,
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
gboolean found;
DC_LOCK (webrtc);
found = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel);
if (found == FALSE) {
GST_FIXME_OBJECT (webrtc, "Received open for unknown data channel");
DC_UNLOCK (webrtc);
return;
}
g_ptr_array_add (webrtc->priv->data_channels, gst_object_ref (channel));
DC_UNLOCK (webrtc);
gst_webrtc_bin_update_sctp_priority (webrtc);
@ -1976,12 +1985,14 @@ _on_data_channel_ready_state (WebRTCDataChannel * channel,
} else if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
gboolean found;
DC_LOCK (webrtc);
found = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel)
|| g_ptr_array_remove (webrtc->priv->data_channels, channel);
if (found == FALSE) {
GST_FIXME_OBJECT (webrtc, "Received close for unknown data channel");
}
DC_UNLOCK (webrtc);
}
}
@ -1996,7 +2007,7 @@ _on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1)
return;
PC_LOCK (webrtc);
DC_LOCK (webrtc);
channel = _find_data_channel_for_id (webrtc, stream_id);
if (!channel) {
channel = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, NULL);
@ -2013,6 +2024,7 @@ _on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
g_ptr_array_add (webrtc->priv->pending_data_channels, channel);
}
DC_UNLOCK (webrtc);
g_signal_connect (channel, "notify::ready-state",
G_CALLBACK (_on_data_channel_ready_state), webrtc);
@ -2022,7 +2034,6 @@ _on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
GST_WARNING_OBJECT (channel, "Failed to link sctp pad %s with channel %"
GST_PTR_FORMAT, GST_PAD_NAME (pad), channel);
gst_object_unref (sink_pad);
PC_UNLOCK (webrtc);
}
static void
@ -2036,9 +2047,9 @@ _on_sctp_state_notify (GstWebRTCSCTPTransport * sctp, GParamSpec * pspec,
if (state == GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED) {
int i;
PC_LOCK (webrtc);
GST_DEBUG_OBJECT (webrtc, "SCTP association established");
DC_LOCK (webrtc);
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
WebRTCDataChannel *channel;
@ -2049,7 +2060,7 @@ _on_sctp_state_notify (GstWebRTCSCTPTransport * sctp, GParamSpec * pspec,
if (!channel->parent.negotiated && !channel->opened)
webrtc_data_channel_start_negotiation (channel);
}
PC_UNLOCK (webrtc);
DC_UNLOCK (webrtc);
}
}
@ -2152,7 +2163,6 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
if (!webrtc->priv->data_channel_transport) {
TransportStream *stream;
GstWebRTCSCTPTransport *sctp_transport;
int i;
stream = _find_transport_for_session (webrtc, session_id);
@ -2201,14 +2211,6 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
GST_ELEMENT (stream->send_bin), "data_sink"))
g_warn_if_reached ();
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
WebRTCDataChannel *channel;
channel = g_ptr_array_index (webrtc->priv->data_channels, i);
webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport);
}
gst_element_sync_state_with_parent (GST_ELEMENT (stream->send_bin));
gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
@ -2226,6 +2228,7 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
}
webrtc->priv->sctp_transport = sctp_transport;
gst_webrtc_bin_update_sctp_priority (webrtc);
}
@ -4502,6 +4505,7 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
remote_port, NULL);
}
DC_LOCK (webrtc);
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
WebRTCDataChannel *channel;
@ -4519,6 +4523,7 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
webrtc_data_channel_start_negotiation (channel);
}
}
DC_UNLOCK (webrtc);
stream->active = TRUE;
@ -5650,6 +5655,7 @@ gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
return NULL;
PC_LOCK (webrtc);
DC_LOCK (webrtc);
/* check if the id has been used already */
if (id != -1) {
WebRTCDataChannel *channel = _find_data_channel_for_id (webrtc, id);
@ -5657,6 +5663,7 @@ gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
GST_ELEMENT_WARNING (webrtc, LIBRARY, SETTINGS,
("Attempting to add a data channel with a duplicate ID: %i", id),
NULL);
DC_UNLOCK (webrtc);
PC_UNLOCK (webrtc);
return NULL;
}
@ -5669,6 +5676,7 @@ gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
if (id == -1) {
GST_ELEMENT_WARNING (webrtc, RESOURCE, NOT_FOUND,
("%s", "Failed to generate an identifier for a data channel"), NULL);
DC_UNLOCK (webrtc);
PC_UNLOCK (webrtc);
return NULL;
}
@ -5679,25 +5687,31 @@ gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
"max-retransmits", max_retransmits, "protocol", protocol,
"negotiated", negotiated, "id", id, "priority", priority, NULL);
if (ret) {
gst_bin_add (GST_BIN (webrtc), ret->appsrc);
gst_bin_add (GST_BIN (webrtc), ret->appsink);
if (!ret) {
DC_UNLOCK (webrtc);
PC_UNLOCK (webrtc);
return ret;
}
gst_element_sync_state_with_parent (ret->appsrc);
gst_element_sync_state_with_parent (ret->appsink);
gst_bin_add (GST_BIN (webrtc), ret->appsrc);
gst_bin_add (GST_BIN (webrtc), ret->appsink);
ret = gst_object_ref (ret);
ret->webrtcbin = webrtc;
g_ptr_array_add (webrtc->priv->data_channels, ret);
gst_webrtc_bin_update_sctp_priority (webrtc);
webrtc_data_channel_link_to_sctp (ret, webrtc->priv->sctp_transport);
if (webrtc->priv->sctp_transport &&
webrtc->priv->sctp_transport->association_established
&& !ret->parent.negotiated) {
webrtc_data_channel_start_negotiation (ret);
} else {
_update_need_negotiation (webrtc);
}
gst_element_sync_state_with_parent (ret->appsrc);
gst_element_sync_state_with_parent (ret->appsink);
ret = gst_object_ref (ret);
ret->webrtcbin = webrtc;
g_ptr_array_add (webrtc->priv->data_channels, ret);
DC_UNLOCK (webrtc);
gst_webrtc_bin_update_sctp_priority (webrtc);
webrtc_data_channel_link_to_sctp (ret, webrtc->priv->sctp_transport);
if (webrtc->priv->sctp_transport &&
webrtc->priv->sctp_transport->association_established
&& !ret->parent.negotiated) {
webrtc_data_channel_start_negotiation (ret);
} else {
_update_need_negotiation (webrtc);
}
PC_UNLOCK (webrtc);
@ -6756,6 +6770,7 @@ gst_webrtc_bin_finalize (GObject * object)
gst_webrtc_session_description_free (webrtc->priv->last_generated_offer);
webrtc->priv->last_generated_offer = NULL;
g_mutex_clear (DC_GET_LOCK (webrtc));
g_mutex_clear (ICE_GET_LOCK (webrtc));
g_mutex_clear (PC_GET_LOCK (webrtc));
g_cond_clear (PC_GET_COND (webrtc));
@ -7219,6 +7234,7 @@ gst_webrtc_bin_init (GstWebRTCBin * webrtc)
g_cond_init (PC_GET_COND (webrtc));
g_mutex_init (ICE_GET_LOCK (webrtc));
g_mutex_init (DC_GET_LOCK (webrtc));
webrtc->rtpbin = _create_rtpbin (webrtc);
gst_bin_add (GST_BIN (webrtc), webrtc->rtpbin);

3
ext/webrtc/gstwebrtcbin.h
View file

@ -100,6 +100,9 @@ struct _GstWebRTCBinPrivate
/* list of data channels we've received a sctp stream for but no data
* channel protocol for */
GPtrArray *pending_data_channels;
/* dc_lock protects data_channels and pending_data_channels */
/* lock ordering is pc_lock first, then dc_lock */
GMutex dc_lock;
guint jb_latency;