Olivier Crête
ee0124cb36
webrtc: Remove the webrtc-priv.h header from public headers
...
And this time for real, also import it in a couple more places
inside the webrtc element to make it build.
Fixes #1607
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2359 >
2021-06-28 16:06:59 +00:00
Olivier Crête
5233c349e7
webrtc lib: Make the rtpreceiver 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
52c676546d
webrtc: Also remove rtcp_transport from the structure
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765 >
2020-11-24 01:59:55 +00:00
Olivier Crête
c5d76d944e
webrtc: Remove APIs to set transport on sender/receiver
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They're not not used ever.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765 >
2020-11-24 01:59:55 +00:00
Olivier Crête
5d5417f271
webrtc: Remove non rtcp-mux code
...
RTCP mux is now always required by the WebRTC spec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765 >
2020-11-24 01:59:55 +00:00
Olivier Crête
78c687da3e
webrtc: Document more objects
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707 >
2020-10-30 16:23:10 -04:00
Sebastian Dröge
f12265d9c5
Revert "webrtc: Document more objects"
...
This reverts commit ad68c6b1eb
.
It breaks the CI until the C# bindings are fixed.
2020-10-08 18:52:50 +03:00
Olivier Crête
ad68c6b1eb
webrtc: Document more objects
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425 >
2020-10-06 16:49:08 -04:00
Niels De Graef
7af1a4566f
Use G_DEFINE_AUTOPTR_CLEANUP_FUNC unconditionally
...
Since we started depending on GLib 2.44, we can be sure this macro is
defined (it will be a no-op on compilers that don't support it).
2019-06-05 08:12:10 +02:00
Thibault Saunier
7fe3f36ac8
Minor documentation fixes
2019-05-13 11:36:27 -04:00
Niels De Graef
39c8c206be
webrtc: Add g_autoptr() support for public types
2019-05-08 15:47:06 +02:00
Maciej Wolny
465ea32d73
webrtc: Remove duplicate declarations
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This causes 'redefinition of typedef ...' errors on GCC 4.5.3
2018-11-28 12:24:37 +00:00
Sebastian Dröge
950ead9215
webrtc: Add some locks to setters and remove non-existing functions from headers
...
https://bugzilla.gnome.org/show_bug.cgi?id=794363
2018-03-16 10:37:24 +02:00
Tim-Philipp Müller
333f636555
webrtc: GST_EXPORT -> GST_WEBRTC_API
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We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:36:33 +00:00
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