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607ef6db60
GstWebRTCSCTPTransport is now made into into an abstract base class that only contains property specifications matching the RTCSctpTransport interface of the W3C WebRTC specification, see https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This class is put into the WebRTC library to expose it for applications and to allow for generation of bindings for non-dynamic languages using GObject introspection. The actual implementation is moved to the subclass WebRTCSCTPTransport located in the WebRTC plugin. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
315 lines
9.1 KiB
C
315 lines
9.1 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_PRIV_H__
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#define __GST_WEBRTC_PRIV_H__
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#include <gst/gst.h>
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#include <gst/webrtc/webrtc_fwd.h>
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#include <gst/webrtc/rtpsender.h>
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#include <gst/webrtc/rtpreceiver.h>
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G_BEGIN_DECLS
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/**
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* GstWebRTCRTPTransceiver:
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* @mline: the mline number this transceiver corresponds to
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* @mid: The media ID of the m-line associated with this
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* transceiver. This association is established, when possible,
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* whenever either a local or remote description is applied. This
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* field is NULL if neither a local or remote description has been
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* applied, or if its associated m-line is rejected by either a remote
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* offer or any answer.
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* @stopped: Indicates whether or not sending and receiving using the paired
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* #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
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* either due to SDP offer/answer
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* @sender: The #GstWebRTCRTPSender object responsible sending data to the
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* remote peer
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* @receiver: The #GstWebRTCRTPReceiver object responsible for receiver data from
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* the remote peer.
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* @direction: The transceiver's desired direction.
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* @current_direction: The transceiver's current direction (read-only)
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* @codec_preferences: A caps representing the codec preferences (read-only)
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* @kind: Type of media (Since: 1.20)
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*
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* Mostly matches the WebRTC RTCRtpTransceiver interface.
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*/
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/**
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* GstWebRTCRTPTransceiver.kind:
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*
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* Type of media
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*
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* Since: 1.20
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*/
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struct _GstWebRTCRTPTransceiver
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{
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GstObject parent;
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guint mline;
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gchar *mid;
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gboolean stopped;
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GstWebRTCRTPSender *sender;
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GstWebRTCRTPReceiver *receiver;
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GstWebRTCRTPTransceiverDirection direction;
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GstWebRTCRTPTransceiverDirection current_direction;
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GstCaps *codec_preferences;
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GstWebRTCKind kind;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCRTPTransceiverClass
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{
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GstObjectClass parent_class;
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/* FIXME; reset */
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gpointer _padding[GST_PADDING];
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};
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/**
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* GstWebRTCRTPSender:
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* @transport: The transport for RTP packets
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* @send_encodings: Unused
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* @priority: The priority of the stream (Since: 1.20)
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*
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* An object to track the sending aspect of the stream
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*
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* Mostly matches the WebRTC RTCRtpSender interface.
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*/
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/**
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* GstWebRTCRTPSender.priority:
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*
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* The priority of the stream
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*
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* Since: 1.20
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*/
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struct _GstWebRTCRTPSender
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{
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GstObject parent;
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/* The MediStreamTrack is represented by the stream and is output into @transport as necessary */
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GstWebRTCDTLSTransport *transport;
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GArray *send_encodings;
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GstWebRTCPriorityType priority;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCRTPSenderClass
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{
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GstObjectClass parent_class;
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void);
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/**
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* GstWebRTCRTPReceiver:
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* @transport: The transport for RTP packets
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*
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* An object to track the receiving aspect of the stream
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*
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* Mostly matches the WebRTC RTCRtpReceiver interface.
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*/
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struct _GstWebRTCRTPReceiver
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{
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GstObject parent;
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/* The MediStreamTrack is represented by the stream and is output into @transport as necessary */
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GstWebRTCDTLSTransport *transport;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCRTPReceiverClass
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{
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GstObjectClass parent_class;
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
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/**
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* GstWebRTCICETransport:
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*/
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struct _GstWebRTCICETransport
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{
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GstObject parent;
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GstWebRTCICERole role;
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GstWebRTCICEComponent component;
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GstWebRTCICEConnectionState state;
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GstWebRTCICEGatheringState gathering_state;
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/* Filled by subclasses */
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GstElement *src;
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GstElement *sink;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCICETransportClass
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{
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GstObjectClass parent_class;
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gboolean (*gather_candidates) (GstWebRTCICETransport * transport);
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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void gst_webrtc_ice_transport_connection_state_change (GstWebRTCICETransport * ice,
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GstWebRTCICEConnectionState new_state);
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GST_WEBRTC_API
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void gst_webrtc_ice_transport_gathering_state_change (GstWebRTCICETransport * ice,
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GstWebRTCICEGatheringState new_state);
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GST_WEBRTC_API
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void gst_webrtc_ice_transport_selected_pair_change (GstWebRTCICETransport * ice);
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GST_WEBRTC_API
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void gst_webrtc_ice_transport_new_candidate (GstWebRTCICETransport * ice, guint stream_id, GstWebRTCICEComponent component, gchar * attr);
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/**
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* GstWebRTCDTLSTransport:
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*/
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struct _GstWebRTCDTLSTransport
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{
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GstObject parent;
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GstWebRTCICETransport *transport;
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GstWebRTCDTLSTransportState state;
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gboolean client;
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guint session_id;
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GstElement *dtlssrtpenc;
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GstElement *dtlssrtpdec;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCDTLSTransportClass
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{
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GstObjectClass parent_class;
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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GstWebRTCDTLSTransport * gst_webrtc_dtls_transport_new (guint session_id);
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GST_WEBRTC_API
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void gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
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GstWebRTCICETransport * ice);
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#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
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#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
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/**
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* GstWebRTCDataChannel:
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*
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* Since: 1.18
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*/
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struct _GstWebRTCDataChannel
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{
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GObject parent;
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GMutex lock;
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gchar *label;
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gboolean ordered;
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guint max_packet_lifetime;
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guint max_retransmits;
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gchar *protocol;
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gboolean negotiated;
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gint id;
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GstWebRTCPriorityType priority;
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GstWebRTCDataChannelState ready_state;
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guint64 buffered_amount;
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guint64 buffered_amount_low_threshold;
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gpointer _padding[GST_PADDING];
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};
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/**
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* GstWebRTCDataChannelClass:
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*
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* Since: 1.18
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*/
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struct _GstWebRTCDataChannelClass
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{
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GObjectClass parent_class;
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void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data);
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void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
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void (*close) (GstWebRTCDataChannel * channel);
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
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/**
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* GstWebRTCSCTPTransport:
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*
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* Since: 1.20
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*/
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struct _GstWebRTCSCTPTransport
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{
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GstObject parent;
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};
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/**
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* GstWebRTCSCTPTransportClass:
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*
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* Since: 1.20
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*/
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struct _GstWebRTCSCTPTransportClass
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{
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GstObjectClass parent_class;
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};
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G_END_DECLS
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#endif /* __GST_WEBRTC_PRIV_H__ */
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