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gstreamer/subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc.h

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webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
/* 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_WEBRTC_H__
#define __GST_WEBRTC_WEBRTC_H__
#include <gst/gst.h>
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/webrtc-enumtypes.h>
#include <gst/webrtc/dtlstransport.h>
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
#include <gst/webrtc/ice.h>
#include <gst/webrtc/icestream.h>
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
#include <gst/webrtc/icetransport.h>
#include <gst/webrtc/rtcsessiondescription.h>
#include <gst/webrtc/rtpreceiver.h>
#include <gst/webrtc/rtpsender.h>
#include <gst/webrtc/rtptransceiver.h>
webrtc: Add GstWebRTCDataChannel to the library API This makes it more discoverable for bindings and allows bindings to generate static API for the signals and functions. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1168 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1313>
2020-06-01 11:46:03 +00:00
#include <gst/webrtc/datachannel.h>
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
#endif /* __GST_WEBRTC_WEBRTC_H__ */
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