/* GStreamer * Copyright (C) 2017 Matthew Waters * * 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_PRIV_H__ #define __GST_WEBRTC_PRIV_H__ #include #include #include #include G_BEGIN_DECLS /** * GstWebRTCRTPTransceiver: * @mline: the mline number this transceiver corresponds to * @mid: The media ID of the m-line associated with this * transceiver. This association is established, when possible, * whenever either a local or remote description is applied. This * field is NULL if neither a local or remote description has been * applied, or if its associated m-line is rejected by either a remote * offer or any answer. * @stopped: Indicates whether or not sending and receiving using the paired * #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled, * either due to SDP offer/answer * @sender: The #GstWebRTCRTPSender object responsible sending data to the * remote peer * @receiver: The #GstWebRTCRTPReceiver object responsible for receiver data from * the remote peer. * @direction: The transceiver's desired direction. * @current_direction: The transceiver's current direction (read-only) * @codec_preferences: A caps representing the codec preferences (read-only) * @kind: Type of media (Since: 1.20) * * Mostly matches the WebRTC RTCRtpTransceiver interface. */ /** * GstWebRTCRTPTransceiver.kind: * * Type of media * * Since: 1.20 */ struct _GstWebRTCRTPTransceiver { GstObject parent; guint mline; gchar *mid; gboolean stopped; GstWebRTCRTPSender *sender; GstWebRTCRTPReceiver *receiver; GstWebRTCRTPTransceiverDirection direction; GstWebRTCRTPTransceiverDirection current_direction; GstCaps *codec_preferences; GstWebRTCKind kind; gpointer _padding[GST_PADDING]; }; struct _GstWebRTCRTPTransceiverClass { GstObjectClass parent_class; /* FIXME; reset */ gpointer _padding[GST_PADDING]; }; /** * GstWebRTCRTPSender: * @transport: The transport for RTP packets * @send_encodings: Unused * @priority: The priority of the stream (Since: 1.20) * * An object to track the sending aspect of the stream * * Mostly matches the WebRTC RTCRtpSender interface. */ /** * GstWebRTCRTPSender.priority: * * The priority of the stream * * Since: 1.20 */ struct _GstWebRTCRTPSender { GstObject parent; /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */ GstWebRTCDTLSTransport *transport; GArray *send_encodings; GstWebRTCPriorityType priority; gpointer _padding[GST_PADDING]; }; struct _GstWebRTCRTPSenderClass { GstObjectClass parent_class; gpointer _padding[GST_PADDING]; }; GST_WEBRTC_API GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void); /** * GstWebRTCRTPReceiver: * @transport: The transport for RTP packets * * An object to track the receiving aspect of the stream * * Mostly matches the WebRTC RTCRtpReceiver interface. */ struct _GstWebRTCRTPReceiver { GstObject parent; /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */ GstWebRTCDTLSTransport *transport; gpointer _padding[GST_PADDING]; }; struct _GstWebRTCRTPReceiverClass { GstObjectClass parent_class; gpointer _padding[GST_PADDING]; }; GST_WEBRTC_API GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void); /** * GstWebRTCICETransport: */ struct _GstWebRTCICETransport { GstObject parent; GstWebRTCICERole role; GstWebRTCICEComponent component; GstWebRTCICEConnectionState state; GstWebRTCICEGatheringState gathering_state; /* Filled by subclasses */ GstElement *src; GstElement *sink; gpointer _padding[GST_PADDING]; }; struct _GstWebRTCICETransportClass { GstObjectClass parent_class; gboolean (*gather_candidates) (GstWebRTCICETransport * transport); gpointer _padding[GST_PADDING]; }; GST_WEBRTC_API void gst_webrtc_ice_transport_connection_state_change (GstWebRTCICETransport * ice, GstWebRTCICEConnectionState new_state); GST_WEBRTC_API void gst_webrtc_ice_transport_gathering_state_change (GstWebRTCICETransport * ice, GstWebRTCICEGatheringState new_state); GST_WEBRTC_API void gst_webrtc_ice_transport_selected_pair_change (GstWebRTCICETransport * ice); GST_WEBRTC_API void gst_webrtc_ice_transport_new_candidate (GstWebRTCICETransport * ice, guint stream_id, GstWebRTCICEComponent component, gchar * attr); /** * GstWebRTCDTLSTransport: */ struct _GstWebRTCDTLSTransport { GstObject parent; GstWebRTCICETransport *transport; GstWebRTCDTLSTransportState state; gboolean client; guint session_id; GstElement *dtlssrtpenc; GstElement *dtlssrtpdec; gpointer _padding[GST_PADDING]; }; struct _GstWebRTCDTLSTransportClass { GstObjectClass parent_class; gpointer _padding[GST_PADDING]; }; GST_WEBRTC_API GstWebRTCDTLSTransport * gst_webrtc_dtls_transport_new (guint session_id); GST_WEBRTC_API void gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport, GstWebRTCICETransport * ice); #define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock) #define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock) /** * GstWebRTCDataChannel: * * Since: 1.18 */ struct _GstWebRTCDataChannel { GObject parent; GMutex lock; gchar *label; gboolean ordered; guint max_packet_lifetime; guint max_retransmits; gchar *protocol; gboolean negotiated; gint id; GstWebRTCPriorityType priority; GstWebRTCDataChannelState ready_state; guint64 buffered_amount; guint64 buffered_amount_low_threshold; gpointer _padding[GST_PADDING]; }; /** * GstWebRTCDataChannelClass: * * Since: 1.18 */ struct _GstWebRTCDataChannelClass { GObjectClass parent_class; void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data); void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str); void (*close) (GstWebRTCDataChannel * channel); gpointer _padding[GST_PADDING]; }; GST_WEBRTC_API void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel); GST_WEBRTC_API void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel); GST_WEBRTC_API void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error); GST_WEBRTC_API void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data); GST_WEBRTC_API void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str); GST_WEBRTC_API void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel); G_END_DECLS #endif /* __GST_WEBRTC_PRIV_H__ */