gstreamer/ext/webrtc/gstwebrtcbin.h
Johan Sternerup 4d514abfd6 webrtcbin: Fix deadlock when receiving new sctp stream
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>
2021-05-12 03:02:27 +00:00

165 lines
5.5 KiB
C

/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_WEBRTC_BIN_H__
#define __GST_WEBRTC_BIN_H__
#include <gst/sdp/sdp.h>
#include "fwd.h"
#include "gstwebrtcice.h"
#include "transportstream.h"
G_BEGIN_DECLS
GType gst_webrtc_bin_pad_get_type(void);
#define GST_TYPE_WEBRTC_BIN_PAD (gst_webrtc_bin_pad_get_type())
#define GST_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPad))
#define GST_IS_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN_PAD))
#define GST_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
#define GST_IS_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN_PAD))
#define GST_WEBRTC_BIN_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
typedef struct _GstWebRTCBinPad GstWebRTCBinPad;
typedef struct _GstWebRTCBinPadClass GstWebRTCBinPadClass;
struct _GstWebRTCBinPad
{
GstGhostPad parent;
GstWebRTCRTPTransceiver *trans;
gulong block_id;
GstCaps *received_caps;
};
struct _GstWebRTCBinPadClass
{
GstGhostPadClass parent_class;
};
GType gst_webrtc_bin_get_type(void);
#define GST_TYPE_WEBRTC_BIN (gst_webrtc_bin_get_type())
#define GST_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN,GstWebRTCBin))
#define GST_IS_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN))
#define GST_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
#define GST_IS_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN))
#define GST_WEBRTC_BIN_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
struct _GstWebRTCBin
{
GstBin parent;
GstElement *rtpbin;
GstElement *rtpfunnel;
GstWebRTCSignalingState signaling_state;
GstWebRTCICEGatheringState ice_gathering_state;
GstWebRTCICEConnectionState ice_connection_state;
GstWebRTCPeerConnectionState peer_connection_state;
GstWebRTCSessionDescription *current_local_description;
GstWebRTCSessionDescription *pending_local_description;
GstWebRTCSessionDescription *current_remote_description;
GstWebRTCSessionDescription *pending_remote_description;
GstWebRTCBundlePolicy bundle_policy;
GstWebRTCICETransportPolicy ice_transport_policy;
GstWebRTCBinPrivate *priv;
};
struct _GstWebRTCBinClass
{
GstBinClass parent_class;
};
struct _GstWebRTCBinPrivate
{
guint max_sink_pad_serial;
gboolean bundle;
GPtrArray *transceivers;
GPtrArray *transports;
GPtrArray *data_channels;
/* 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;
GstWebRTCSCTPTransport *sctp_transport;
TransportStream *data_channel_transport;
GstWebRTCICE *ice;
GArray *ice_stream_map;
GMutex ice_lock;
GArray *pending_remote_ice_candidates;
GArray *pending_local_ice_candidates;
/* peerconnection variables */
gboolean is_closed;
gboolean need_negotiation;
/* peerconnection helper thread for promises */
GMainContext *main_context;
GMainLoop *loop;
GThread *thread;
GMutex pc_lock;
GCond pc_cond;
gboolean running;
gboolean async_pending;
GList *pending_pads;
GList *pending_sink_transceivers;
/* count of the number of media streams we've offered for uniqueness */
/* FIXME: overflow? */
guint media_counter;
/* the number of times create_offer has been called for the version field */
guint offer_count;
GstWebRTCSessionDescription *last_generated_offer;
GstWebRTCSessionDescription *last_generated_answer;
gboolean tos_attached;
};
typedef GstStructure *(*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data);
typedef struct
{
GstWebRTCBin *webrtc;
GstWebRTCBinFunc op;
gpointer data;
GDestroyNotify notify;
GstPromise *promise;
} GstWebRTCBinTask;
gboolean gst_webrtc_bin_enqueue_task (GstWebRTCBin * pc,
GstWebRTCBinFunc func,
gpointer data,
GDestroyNotify notify,
GstPromise *promise);
G_END_DECLS
#endif /* __GST_WEBRTC_BIN_H__ */