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/* 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__ */
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