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Author SHA1 Message Date
Johan Sternerup
607ef6db60 webrtc: Split sctptransport into lib and implementation parts 
GstWebRTCSCTPTransport is now made into into an abstract base class
that only contains property specifications matching the
RTCSctpTransport interface of the W3C WebRTC specification, see
https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This
class is put into the WebRTC library to expose it for applications and
to allow for generation of bindings for non-dynamic languages using
GObject introspection.

The actual implementation is moved to the subclass WebRTCSCTPTransport
located in the WebRTC plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
 2021-08-25 13:20:22 +00:00
Sebastian Dröge
0e559fc2f3 webrtcbin: Sync to the clock per stream and not per bundle 
By using the clocksync inside the dtlssrtpenc, all streams inside a
bundled are synchronized together. This will cause problems if their
buffers are not already arriving synchronized: clocksync would wait for
a buffer on one stream and then buffers from the other stream(s) with
lower timestamps would all be sent out too late.

Placing the clocksync before the rtpbin and rtpfunnel synchronizes each
stream individually and they will be send out more smoothly as a result.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2355>
 2021-06-28 16:38:33 +00:00
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
Sebastian Dröge
03d3e0fe73 webrtc: Re-add WebRTC object docs to the public headers 
So they end up in the generated documentation and the Since markers
appear in the .gir files too.

Also remove wrong "Since: 1.16" markers for some objects that were
available since 1.14.0 already.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1609

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2366>
 2021-06-28 14:45:37 +00:00
Olivier Crête
a931e31141 webrtc lib: Make the datachannel 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
dd2da6f2b4 webrtc lib: Make the DTLSTransport 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
a0813c5bd2 webrtc lib: Make the icetransport 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
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
a6593753a5 webrtc lib: Make the rtpsender 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
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
Marijn Suijten
061e32b197 Add @ prefix to enum-variant references in documentation 
Found while working on GStreamer-rs documentation, some enums had this
bit of text pasted verbatim in the enum documentation rather than
attached to the enum-variant.  Fortunately it seems these in WebRTC and
D3D11 are the only ones matching the non-@-prefixed pattern:

    ^ \* GST_\w+:\s*\w+

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2118>
 2021-03-28 13:08:24 +00:00
Mathieu Duponchelle
86c009e7aa webrtc: expose transport property on sender and receiver 
As advised by !1366#note_629558 , the nice transport should be
accessed through:

> transceiver->sender/receiver->transport/rtcp_transport->icetransport

All the objects on the path can be accessed through properties
except sender/receiver->transport. This patch addresses that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1952>
 2021-01-13 19:22:42 +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 
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
cca313ecd8 rtpsender: Add API to set the priority 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:24:40 -04:00
Olivier Crête
7be09a5f22 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
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
Xavier Claessens
2efb4a7adb Meson: Use pkg-config generator 2020-10-23 11:14:18 -04:00
Sebastian Dröge
cc7e98816f Revert "webrtc: Save the media kind in the transceiver" 
This reverts commit f54d8e9945.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:12 +03:00
Sebastian Dröge
a40d6f4994 Revert "rtpsender: Add API to set the priority" 
This reverts commit a8b287c764.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:10 +03:00
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
Sebastian Dröge
74a42c5ba8 Revert "webrtc: Add hotdoc style since tags" 
This reverts commit 63a5fa818c.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:51:37 +03:00
Olivier Crête
63a5fa818c webrtc: Add hotdoc style since tags 
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 
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 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
f54d8e9945 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Matthew Waters
7489addc0a webrtc/datachannel: free previous protocol/label fields 
Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Sebastian Dröge
ab82893941 webrtc: Add Since: 1.18 markers to the new datachannel library API 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1315>
 2020-06-03 10:32:00 +03:00
Sebastian Dröge
b25d153c34 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>
 2020-06-02 21:04:37 +00:00
Jan Schmidt
8e3472faee webrtc: Use the dtlssrtenc rtp-sync property 
Instead of synchronising at the ICE transport, do clock sync for the
RTP stream at the DTLS transport via the dtlssrtpenc rtp-sync
property. This avoids delaying RTCP while waiting until it is time
to output an RTP packet when rtcp-mux is enabled.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1212
 2020-02-27 12:30:32 +00:00
Sebastian Dröge
b2e7739364 webrtc/dtlstransport: Proxy DTLS connection state from the DTLS elements to the transport 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
 2020-01-19 11:16:34 +00:00
Niels De Graef
d8f61515d8 Don't pass default GLib marshallers for signals 
By passing NULL to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.

Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
 2019-11-06 14:27:46 +00:00
Tim-Philipp Müller
f218ec2794 Remove autotools build system 2019-10-14 13:54:27 +01:00
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
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
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
Sebastian Dröge
b3bf3a0d21 webrtc: Add various Since markers to new types after 1.14.0 2019-05-21 12:16:31 +03:00
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
Mathieu Duponchelle
85c75bb23b webrtc: expose ice-transport-policy property 
This is the equivalent of iceTransportPolicy in the RTCConfiguration
dictionary.

Only two values are implemented:

* all: default behaviour
* relay: only gather relay candidates

The third member of the iceTransportPolicy enum, "public", is
obsolete.
 2019-01-23 22:47:51 +00:00
Maciej Wolny
465ea32d73 webrtc: Remove duplicate declarations 
This causes 'redefinition of typedef ...' errors on GCC 4.5.3
 2018-11-28 12:24:37 +00:00
Mathieu Duponchelle
9f684a2f81 webrtcbin: implement support for group: BUNDLE 2018-10-15 14:17:35 +02:00