| /* 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" |
| |
| G_BEGIN_DECLS |
| |
| #define GST_WEBRTC_BIN_ERROR gst_webrtc_bin_error_quark () |
| GQuark gst_webrtc_bin_error_quark (void); |
| |
| typedef enum |
| { |
| GST_WEBRTC_BIN_ERROR_FAILED, |
| GST_WEBRTC_BIN_ERROR_INVALID_SYNTAX, |
| GST_WEBRTC_BIN_ERROR_INVALID_MODIFICATION, |
| GST_WEBRTC_BIN_ERROR_INVALID_STATE, |
| GST_WEBRTC_BIN_ERROR_BAD_SDP, |
| GST_WEBRTC_BIN_ERROR_FINGERPRINT, |
| } GstWebRTCJSEPSDPError; |
| |
| 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; |
| |
| guint mlineindex; |
| |
| GstWebRTCRTPTransceiver *trans; |
| }; |
| |
| 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; |
| |
| 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; |
| |
| GstWebRTCBinPrivate *priv; |
| }; |
| |
| struct _GstWebRTCBinClass |
| { |
| GstBinClass parent_class; |
| }; |
| |
| struct _GstWebRTCBinPrivate |
| { |
| guint max_sink_pad_serial; |
| |
| gboolean bundle; |
| GArray *transceivers; |
| GArray *session_mid_map; |
| GArray *transports; |
| |
| GstWebRTCICE *ice; |
| GArray *ice_stream_map; |
| GArray *pending_ice_candidates; |
| |
| /* peerconnection variables */ |
| gboolean is_closed; |
| gboolean need_negotiation; |
| gpointer sctp_transport; /* FIXME */ |
| |
| /* 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; |
| |
| /* count of the number of media streams we've offered for uniqueness */ |
| /* FIXME: overflow? */ |
| guint media_counter; |
| |
| GstStructure *stats; |
| }; |
| |
| typedef void (*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data); |
| |
| typedef struct |
| { |
| GstWebRTCBin *webrtc; |
| GstWebRTCBinFunc op; |
| gpointer data; |
| GDestroyNotify notify; |
| // GstPromise *promise; /* FIXME */ |
| } GstWebRTCBinTask; |
| |
| void gst_webrtc_bin_enqueue_task (GstWebRTCBin * pc, |
| GstWebRTCBinFunc func, |
| gpointer data, |
| GDestroyNotify notify); |
| |
| G_END_DECLS |
| |
| #endif /* __GST_WEBRTC_BIN_H__ */ |