Commit graph

33 commits

Author SHA1 Message Date
Matthew Waters
e463bcfadf tests/webrtc: check for more sdp things across the board
e.g.

- test for a=setup:$val and direction attributes in all tests
- test number of media sections
- test number of formats in each m= section (for audio/video)
- test no duplicate formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2093>
2021-03-19 18:02:21 +11:00
Matthew Waters
2bed220771 webrtc: don't generate duplicate rtx payloads when bundle-policy is set
It was possible to generate a SDP that had an RTX payload type
that matched one of the media payload types when providing caps via
codec_preferences without any sink pads.

Fixes

m=video 9 UDP/TLS/RTP/SAVPF 96
...
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack pli
a=fmtp:96 apt=96

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
2021-03-09 02:22:35 +00:00
Ilya Kreymer
92626535c7 webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
2021-03-01 14:42:17 +00:00
Matthew Waters
e2d88f0569 webrtc: propagate more errors through the promise
Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>
2020-09-14 04:04:29 +00:00
Matthew Waters
e4b848e2a8 tests/webrtc: unref GBytes after use
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
2020-08-24 17:02:35 +10:00
Matthew Waters
02c8e66ff1 webrtc: fix an off-by-one calculating low-threshold
We were not signalling low-threshold when the previous amount was at
exactly the low-threshold mark.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>
2020-05-06 15:49:58 +10:00
Matthew Waters
d0b20f8bba webrtc: fix a slightly racy test
There is no guarantee that the peer data channel has transitioned to
open when we do.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>
2020-05-06 15:49:58 +10:00
Matthew Waters
d0be6b74f2 tests/webrtc: fix a data channel leak in a test
test_data_channel_create_after_negotiate overrides the data_channel_data
without ever freeing it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
7aa58954c1 tests/webrtc: move bus thread creation earlier
Fixes a small deadlock race where the bus watch GSource could execute before
the unlock mutex GSource.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Stéphane Cerveau
6bc0e9527e remove various useless linefeed in logs 2019-12-11 10:51:29 +01:00
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
Matthew Waters
62cc5e51d1 tests/webrtc: wait until the SDP has been set before continuing
If we renegotiate, then it is currently possible for an added stream to
be added to webrtcbin before the SDP is complete.  This causes an
internal inconsistency as there is a 'pending sink transceiver' without
a corresponding media section in the sdp.  It also does not have an
associated transport stream and will fail in _connect_input_stream().
2019-05-30 21:33:09 +10:00
Matthew Waters
979daea7f2 tests/webrtc: fix racy test with a prenegotiated data channel
If both data channels become ready simultaneously, then the two integer
read-add-update cycles can execute concurrently and only ever increment
once instead of the required twice.  Use an atomic add instead.
2019-05-30 21:33:09 +10:00
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
015cb75f66 tests/webrtc: a couple of debug/error string fixes 2019-05-30 21:33:09 +10:00
Matthew Waters
be35735989 tests/webrtc: rewrite bundle checks for separate validate_sdp passes
Improves reusability
2019-05-30 21:33:09 +10:00
Matthew Waters
2bb1fde47c tests/webrtc: add helper for getting the offer/answer element 2019-05-30 21:33:09 +10:00
Matthew Waters
b48e2947bf tests/webrtc: only check audio/video for direction attributes 2019-05-30 21:33:09 +10:00
Matthew Waters
bd92b2f7c4 webrtc: fix answer creation with multiple streams and similar caps 2019-05-30 21:26:46 +10:00
Matthew Waters
ebb9c3c298 tests/webrtc: factor out sdp validation into a single function 2019-05-30 21:26:46 +10:00
Matthew Waters
eb79f95bf8 tests/webrtc: validate number of sdp media using validate_sdp 2019-05-30 21:26:46 +10:00
Matthew Waters
7e1cdbfd4d tests/webrtc: allow multiple validation functions 2019-05-30 21:26:46 +10:00
Matthew Waters
120a40cf25 tests/webrtc: test that duplicate negotiations succeed 2019-05-30 21:26:46 +10:00
Matthew Waters
65c9c4b406 tests/webrtc: fix tests for no libnice
webrtcbin will fail the state change to READY when libnice elements are
not available.
2018-11-26 17:13:08 +11:00
Matthew Waters
57a006d8a5 tests/webrtc: use the existing functions in the plugin
Instead of redefining our own, use the function implementations in
webrtcsdp.c and utils.c
2018-11-26 17:13:08 +11:00
Mathieu Duponchelle
9f684a2f81 webrtcbin: implement support for group: BUNDLE 2018-10-15 14:17:35 +02:00
Matthew Waters
21bf3a35ac webrtc/datachannel: fix support for prenegotiated channels
With prenegotiated channels, the data-channel protocol is not used and
instead the channel's negotiation is intended to be performed out of band in
some application-specific manner.

Comes with test!
2018-10-09 02:38:14 +11:00
Matthew Waters
7bf18ad258 webrtc: start in the closed state
This means that we will reject all operations before we've transitioned
into READY.

This also fixes the tests using the default GMainContext in the NULL
state instead of the webrtcbin internal GMainContext and thread.  Also
removes a potential ordering race where on the element transitioning to
READY, an operations could have been queued on two different threads and
removing a guarentee on operation ordering.
2018-10-08 21:56:31 +11:00
Matthew Waters
faf212cd3a tests/webrtcbin: add a couple more dependant element checks
Without the sctp elements, all the data channel tests will fail
2018-10-04 16:44:55 +10:00
Matthew Waters
07e9374eff webrtcbin: add support for data channels based on SCTP
Mostly follows the W3C specification
https://www.w3.org/TR/webrtc/#peer-to-peer-data-api

With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=794351
2018-09-21 19:45:12 +10:00
Jan Schmidt
0af199e25e webrtc: Add some ssrc to caps in the unit test.
The unit test uses incomplete caps to test webrtcbin,
causing some weirdness generating RTX stream mappings.

https://bugzilla.gnome.org/show_bug.cgi?id=796810
2018-07-15 23:07:37 +10:00
Mathieu Duponchelle
5c450c5992 webrtcbin: implement support for FEC and RTX
https://bugzilla.gnome.org/show_bug.cgi?id=795044
2018-05-09 14:46:14 +02:00
Matthew Waters
1894293d63 webrtcbin: an element that handles the transport aspects of webrtc connections
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