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
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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
cca313ecd8
rtpsender: Add API to set the priority
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707 >
2020-10-30 16:24:40 -04:00
Olivier Crête
78c687da3e
webrtc: Document more objects
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707 >
2020-10-30 16:23:10 -04:00
Sebastian Dröge
a40d6f4994
Revert "rtpsender: Add API to set the priority"
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This reverts commit a8b287c764
2020-10-08 18:53:10 +03:00
Sebastian Dröge
f12265d9c5
Revert "webrtc: Document more objects"
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This reverts commit ad68c6b1eb
2020-10-08 18:52:50 +03:00
Sebastian Dröge
74a42c5ba8
Revert "webrtc: Add hotdoc style since tags"
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This reverts commit 63a5fa818c
2020-10-08 18:51:37 +03:00
Olivier Crête
63a5fa818c
webrtc: Add hotdoc style since tags
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We're stuck having to add a separate comment for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425 >
2020-10-06 16:52:48 -04:00
Olivier Crête
ad68c6b1eb
webrtc: Document more objects
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425 >
2020-10-06 16:49:08 -04:00
Olivier Crête
a8b287c764
rtpsender: Add API to set the priority
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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
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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
Sebastian Dröge
b1ca76377f
webrtc: Remove unused parameter from rtpsender constructor
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https://bugzilla.gnome.org/show_bug.cgi?id=794363
2018-03-16 10:37:24 +02:00
Sebastian Dröge
950ead9215
webrtc: Add some locks to setters and remove non-existing functions from headers
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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
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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
