Commit graph

10 commits

Author SHA1 Message Date
Albert Sjölund
6341817018 webrtc: patch leak caused by early return
In webrtc_data_channel_send functions, both data and string,
an early return on a non-open datachannel caused it to leak
the buffer used for pushing to appsrc, meaning any buffer
sent after leaving the open state was leaked in full.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4206>
2023-03-17 12:32:23 +00:00
Matthew Waters
5ca3988420 webrtc/datachannel: handle error messages from appsrc/sink
Fixes a possible race where closing a data channel may produce e.g.
not-linked errors.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3381>
2022-11-11 10:13:27 +00:00
Johan Sternerup
212c09a70e webrtc: return error when sending on non-open datachannel
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.

Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
2022-10-05 11:08:30 +00:00
Philippe Normand
779ca38229 webrtcdatachannel: Chain to parent class constructed
And add a debug log statement.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2427>
2022-06-07 11:29:32 +00:00
Philippe Normand
556ee45bfa datachannel: Notify low buffered amount according to spec
Quoting
https://www.w3.org/TR/webrtc/#dom-rtcdatachannel-bufferedamountlowthreshold

The bufferedAmountLowThreshold attribute sets the threshold at which the
bufferedAmount is considered to be low. When the bufferedAmount decreases from
above this threshold to **equal** or below it, the bufferedamountlow event fires.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2448>
2022-05-19 05:52:51 +00:00
Matthew Waters
5741ee38e0 webrtc/datachannel: fix use-after-free in sctp state notification
g_signal_disconnect*() doesn't stop any existing callbacks from running
which means that if the notify::state callback is in progress in one
thread and the data channel object is finalize()ed in another thread,
then there could be a use-after-free trying lock the data channel
object.

We can't reasonably use a GWeakRef as we don't have a 'parent' object to
free the GWeakRef after the data channel is finalized.  This is also
complicated by the fact that the application can hold a reference to the
data channel object that would live beyond the lifetime of webrtcbin
itself.

We solve this by implementing a ghetto weak-ref solution internally with
a list of outstanding data channels.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Philippe Normand
4254920b72 webrtc: Expose RTCError enum
The error codes not complying with the spec are now notified with the
GST_WEBRTC_ERROR_INTERNAL_FAILURE code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1485>
2022-01-29 14:42:22 +00:00
Sangchul Lee
5cedf017f5 webrtc: Fix memory leaks
Redundant condition and unreachable codes are also removed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1544>
2022-01-22 11:21:18 +00:00
Philippe Normand
f0e6959bba webrtcdatachannel: Notify buffered-amount property updates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1484>
2022-01-02 10:18:35 +00:00
Thibault Saunier
019971a3c7 Move files from gst-plugins-bad into the "subprojects/gst-plugins-bad/" subdir 2021-09-24 16:14:36 -03:00
Renamed from ext/webrtc/webrtcdatachannel.c (Browse further)