mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
315 lines
9.1 KiB
C
315 lines
9.1 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_PRIV_H__
|
|
#define __GST_WEBRTC_PRIV_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/webrtc/webrtc_fwd.h>
|
|
#include <gst/webrtc/rtpsender.h>
|
|
#include <gst/webrtc/rtpreceiver.h>
|
|
|
|
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);
|
|
|
|
|
|
/**
|
|
* GstWebRTCSCTPTransport:
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
struct _GstWebRTCSCTPTransport
|
|
{
|
|
GstObject parent;
|
|
};
|
|
|
|
/**
|
|
* GstWebRTCSCTPTransportClass:
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
struct _GstWebRTCSCTPTransportClass
|
|
{
|
|
GstObjectClass parent_class;
|
|
};
|
|
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_WEBRTC_PRIV_H__ */
|