Commit graph

67 commits

Author SHA1 Message Date
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
Dave Piché
574cbbf0b5 webrtc: fix log error message in function gst_webrtc_bin_set_local_description
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1511>
2022-01-13 15:11:35 +00:00
Olivier Crête
29befed685 webrtcbin: Store the ssrc of the last received packet
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Mathieu Duponchelle
abd61732bf webrtcbin: bind transceiver's fec-percentage to encoder percentage
Allows for dynamic control of the applied FEC overhead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
2021-12-14 17:34:53 +00:00
Mathieu Duponchelle
06893b8b5e webrtcbin: fix ulpfec / red for the BUNDLE case
* Add fec / red encoders as direct children of webrtcbin, instead
  of providing them to rtpbin through the request-fec-encoder signal.

  That is because they need to be placed before the rtpfunnel, which
  is placed upstream of rtpbin.

* Update configuration of red decoders to set a list of RED payloads
  on them, instead of setting the pt property.

  That is because there may be one RED pt per media in the same session.

* Connect to request-fec-decoder-full instead of request-fec-decoder,
  in order to instantiate FEC decoders according to the payload type
  of the stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
2021-12-14 17:34:53 +00:00
Mathieu Duponchelle
e90859f4d8 webrtcbin: deduplicate extmaps
When an extmap is defined twice for the same ID, firefox complains and
errors out (chrome is smart enough to accept strict duplicates).

To work around this, we deduplicate extmap attributes, and also error
out when a different extmap is defined for the same ID.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1383>
2021-11-25 18:38:22 +00:00
Sebastian Dröge
f9a97efbe1 webrtcbin: Clear errors from finding codec preferences before the next iteration
The media is just skipped and the error is not propagated to the caller,
so keeping it around here would cause assertions a bit later when trying
to set a new error over the old one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
2021-11-04 10:51:15 +00:00
Sebastian Dröge
30153f1591 webrtcbin: Move addition of attributes to the caps after making sure they're not empty or any
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
2021-11-04 10:51:15 +00:00
Sebastian Dröge
d628ccf0e5 webrtcbin: Don't require fixed caps when querying caps for a transceiver pad to match it with a media
Upstream caps might for example be
  application/x-rtp,media=audio,encoding-name={OPUS, X-GST-OPUS-DRAFT-SPITTKA-00, multiopus}
and while that is not fixed caps it is enough to match it with a media.

Only caps structures that have the correct structure name and that have
the media and encoding-name field are preserved, but if both are present
then these caps are used as "codec preferences".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
2021-11-04 10:51:15 +00:00
Mathieu Duponchelle
303c8025c6 webrtcbin: fix check_negotiation computing on caps event
It seems logical that check_negotiation be true if received_caps
is *not* equal to the new caps.

Also clean up handling of transceivers' ssrc events, as this
patch triggered a leaky code path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
2021-10-28 19:05:59 +00:00
Mathieu Duponchelle
be0b5c54fd webrtcbin: connect input stream when receiving caps
.. if a current direction has already been set

When `webrtcbin` has created an offer based on codec_preferences,
it might not have received caps on its sinkpads by the time a
remote description is set, in which case we want to connect the
input stream upon actual reception of the caps instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
2021-10-28 19:05:59 +00:00
Mathieu Duponchelle
a9506f20d3 webrtcbin: consider pads with trans->codec_preferences ready
.. when determining whether we can emit on-negotiation-needed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
2021-10-28 19:05:59 +00:00
Rob Agar
641b319fd6 webrtcbin: Also check data channel transport when collating connection state
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/838

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1224>
2021-10-28 05:05:44 +00:00
Rob Agar
66a24023c0 webrtcbin: fix prevention of webrtcbin deletion due to ref held by probe callback
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/810

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1150>
2021-10-18 10:42:12 +01:00
Sebastian Dröge
3011fa7ddd webrtcbin: Use the same promise reply structure name everywhere
This was an inconsistent mix of different names in the past. The name
has no meaning at all so let's set all to "application/x-gst-promise".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1099>
2021-10-09 11:45:46 +03:00
Sebastian Dröge
6d9ca9c679 webrtcbin: Always set SINK/SRC flags
webrtcbin can act as a sink/source depending on the SDP later. Without
setting this here already, surrounding bins might not notice this and
the pipeline configuration might become inconsistent, e.g. with regards
to latency.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/737

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/900>
2021-09-25 16:33:13 +03:00
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/gstwebrtcbin.c (Browse further)