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1894293d63
SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
154 lines
4.9 KiB
C
154 lines
4.9 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_WEBRTC_BIN_H__
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#define __GST_WEBRTC_BIN_H__
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#include <gst/sdp/sdp.h>
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#include "fwd.h"
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#include "gstwebrtcice.h"
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G_BEGIN_DECLS
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#define GST_WEBRTC_BIN_ERROR gst_webrtc_bin_error_quark ()
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GQuark gst_webrtc_bin_error_quark (void);
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typedef enum
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{
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GST_WEBRTC_BIN_ERROR_FAILED,
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GST_WEBRTC_BIN_ERROR_INVALID_SYNTAX,
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GST_WEBRTC_BIN_ERROR_INVALID_MODIFICATION,
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GST_WEBRTC_BIN_ERROR_INVALID_STATE,
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GST_WEBRTC_BIN_ERROR_BAD_SDP,
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GST_WEBRTC_BIN_ERROR_FINGERPRINT,
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} GstWebRTCJSEPSDPError;
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GType gst_webrtc_bin_pad_get_type(void);
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#define GST_TYPE_WEBRTC_BIN_PAD (gst_webrtc_bin_pad_get_type())
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#define GST_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPad))
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#define GST_IS_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN_PAD))
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#define GST_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
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#define GST_IS_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN_PAD))
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#define GST_WEBRTC_BIN_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
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typedef struct _GstWebRTCBinPad GstWebRTCBinPad;
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typedef struct _GstWebRTCBinPadClass GstWebRTCBinPadClass;
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struct _GstWebRTCBinPad
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{
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GstGhostPad parent;
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guint mlineindex;
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GstWebRTCRTPTransceiver *trans;
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};
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struct _GstWebRTCBinPadClass
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{
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GstGhostPadClass parent_class;
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};
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GType gst_webrtc_bin_get_type(void);
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#define GST_TYPE_WEBRTC_BIN (gst_webrtc_bin_get_type())
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#define GST_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN,GstWebRTCBin))
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#define GST_IS_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN))
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#define GST_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
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#define GST_IS_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN))
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#define GST_WEBRTC_BIN_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
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struct _GstWebRTCBin
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{
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GstBin parent;
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GstElement *rtpbin;
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GstWebRTCSignalingState signaling_state;
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GstWebRTCICEGatheringState ice_gathering_state;
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GstWebRTCICEConnectionState ice_connection_state;
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GstWebRTCPeerConnectionState peer_connection_state;
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GstWebRTCSessionDescription *current_local_description;
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GstWebRTCSessionDescription *pending_local_description;
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GstWebRTCSessionDescription *current_remote_description;
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GstWebRTCSessionDescription *pending_remote_description;
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GstWebRTCBinPrivate *priv;
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};
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struct _GstWebRTCBinClass
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{
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GstBinClass parent_class;
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};
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struct _GstWebRTCBinPrivate
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{
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guint max_sink_pad_serial;
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gboolean bundle;
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GArray *transceivers;
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GArray *session_mid_map;
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GArray *transports;
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GstWebRTCICE *ice;
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GArray *ice_stream_map;
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GArray *pending_ice_candidates;
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/* peerconnection variables */
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gboolean is_closed;
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gboolean need_negotiation;
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gpointer sctp_transport; /* FIXME */
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/* peerconnection helper thread for promises */
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GMainContext *main_context;
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GMainLoop *loop;
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GThread *thread;
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GMutex pc_lock;
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GCond pc_cond;
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gboolean running;
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gboolean async_pending;
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GList *pending_pads;
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/* count of the number of media streams we've offered for uniqueness */
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/* FIXME: overflow? */
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guint media_counter;
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GstStructure *stats;
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};
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typedef void (*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data);
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typedef struct
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{
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GstWebRTCBin *webrtc;
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GstWebRTCBinFunc op;
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gpointer data;
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GDestroyNotify notify;
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// GstPromise *promise; /* FIXME */
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} GstWebRTCBinTask;
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void gst_webrtc_bin_enqueue_task (GstWebRTCBin * pc,
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GstWebRTCBinFunc func,
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gpointer data,
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GDestroyNotify notify);
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G_END_DECLS
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#endif /* __GST_WEBRTC_BIN_H__ */
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