mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-20 02:50:16 +00:00
b6114a7fed
The simple case where this breaks is if you add a datachannel and want to add a new pad (a new media) after). Another case where this is broken is if the order of the media is forced to something different by the peer. It's more simple to just split both things completely. In practice, the pads will be named in the order in which they are allocated, so it shouldn't change the current behaviour, just enable new ones. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
163 lines
5.4 KiB
C
163 lines
5.4 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;
|
|
GArray *session_mid_map;
|
|
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;
|
|
|
|
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 void (*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__ */
|