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30 commits

Author SHA1 Message Date
Sangchul Lee
c88a68c411 webrtcbin: Avoid access of freed memory 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2258>
 2022-04-21 10:34:31 +00:00
Sangchul Lee
1ba785b070 webrtcbin: Update documentation of 'get-stats' action signal 
Some stats fields are updated according to the current implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2034>
 2022-03-26 02:13:08 +00:00
Nirbheek Chauhan
3f1f1700bc webrtcbin: Warn when offer didn't intersect with transceiver caps 
We were silently falling back to creating a recvonly offer if the caps
didn't intersect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1995>
 2022-03-20 22:38:15 +00:00
Sangchul Lee
618bbf80dc webrtcbin: Check data channel transport for notifying 'ice-gathering-state' 
Previously, it did not care about data channel's. It is fixed by adding
some conditions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1981>
 2022-03-18 11:31:37 +00:00
Sangchul Lee
12f7bace76 webrtcice: Fix memory leaks in gst_webrtc_ice_add_candidate() 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1704>
 2022-02-15 06:58:31 +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
Jakub Adam
bea8cba5e6 webrtcbin: Chain up to parent constructed method 
Failing to do so makes GstWebRTCBin invisible to the leaks tracer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1587>
 2022-01-27 17:43:18 +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
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
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
Philippe Normand
43856a0735 webrtcstats: Fix null pointer dereference 
If there is no jitterbuffer stats we should not attempt to store them in the
global stats structure.

Also add a g_return_if_fail in _gst_structure_take_structure() about this
because it is a programmer error to pass an invalid pointer address there.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1479>
 2021-12-29 15:55:57 +00:00
Olivier Crête
818a185b5d webrtcstats: Fall back to last packet ssrc if caps dont provide it 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
 2021-12-23 23:48:17 -05:00
Olivier Crête
4e32d6bf3e webrtcstats: Use our own caps instead of the sticky event 
The sticky event seems to get cleared sometimes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
 2021-12-23 23:48:17 -05:00
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
Olivier Crête
fc7e7f5ccc webrtc stats: Remove duplicate structure get 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
 2021-12-23 23:48:17 -05:00
Olivier Crête
f35435f1f7 webrtc stats: Add more details about codecs into the stats 
This makes the output a little closer to what the upstream stats are.

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