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12 commits

Author SHA1 Message Date
Thibault Saunier 019971a3c7 Move files from gst-plugins-bad into the "subprojects/gst-plugins-bad/" subdir 2021-09-24 16:14:36 -03:00
Olivier Crête b5f2de3124 webrtc lib: Make the transceiver struct private 
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
 2021-06-21 20:53:09 +00:00
Olivier Crête e9f14ed117 webrtcbin: Hold lock while accessing the codec preferences 
They could be changed at runtime by the application, so take the lock
when modifying them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Olivier Crête 4c3270409d webrtc rtptransceiver: Implement "codec-preferences" property 
This allows safer access to the internals of the codec-preferences

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Olivier Crête 97a78a903a webrtc rtptransceiver: Implement "kind" property 
Implement the property as read-only to follow the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Olivier Crête 5fd0ee3227 webrtc rtptransceiver: Implement "current-direction" property 
Implement the property as read-only to follow the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Olivier Crête 7e7678f4cb webrtc rtptransceiver: Implement "mid" property 
Implement the property as read-only to follow the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Mathieu Duponchelle 42adb02a10 docstrings: port ulinks to markdown links 2019-08-23 20:14:12 +02:00
Sebastian Dröge 28b0be4036 rtptransceiver: Remove direction setter and vfunc and replace it by a property 
It was changed from a function to a property in the latest WebRTC spec.
 2019-08-06 12:22:21 +00:00
Matthew Waters 177aa22bcd webrtc: Initial support for stream addition/removal 
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
  will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
 2019-05-30 21:33:09 +10:00
Matthew Waters c3c4b07ad3 webrtc/transceiver: add a set_direction function 
Matches the setDirection() from the W3C spec and allows changing the
transceiver direction at the next negotiation cycle.
 2019-05-30 21:33:09 +10:00
Matthew Waters 1894293d63 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
 2018-02-02 15:02:21 +11:00