Commit graph

89 commits

Author SHA1 Message Date
Víctor Manuel Jáquez Leal
5542dd395d jpegparse: Rewrite element.
Now it uses the JPEG parser in libgstcodecparsers, while the whole
code is simplified by relying more in baseparser class for tag
handling.

The element now signals chroma-format and default framerate is 0/1,
which is for still-images.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1473>
2022-05-20 08:51:23 +00:00
Víctor Manuel Jáquez Leal
fa2b697389 tests: jpegparse: Mark data as static.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1473>
2022-05-20 08:51:23 +00:00
Stéphane Cerveau
fcc6fa21e9 srtp: fix flaky unit test
Use different port for each test to avoid other UDP
packet to be received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2294>
2022-04-28 07:53:19 +00:00
Stéphane Cerveau
12776ba0fd srtp: add unit tests
Enable unit tests in meson.build
Add test_play_key_error to check the stats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2027>
2022-04-25 13:57:42 +00:00
Sangchul Lee
a801d6dd63 webrtcstats: Unify 'packets-lost' data type to int64
Previously, 'packets-lost' member of RTCReceivedRtpStreamStats had
a value of G_TYPE_INT from rtpsource or a value of G_TYPE_UINT64
from rtpjitterbuffer.
Because of the negative value of estimated amount of packets lost
in rtpsource as well as the description in
https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats
it is fixed to set this value with G_TYPE_INT64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2049>
2022-03-31 05:37:39 +00:00
Matthew Waters
5bfe36746a webrtc: implement initial simulcast fec/rtx usage
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:41 +00:00
Matthew Waters
831b34fb43 tests/webrtc: fix a use-after-free in test_data_channel_close
g_object_weak_ref() is not thread-safe and the data channel object's
refs/unrefs can happen on multiple threads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
f11e0e76c6 tests/webrtc: fix a race in the tests related to state tracking
If things progress fast enough, some state changes may not be seen be
the waiting code.

Fix by:
1. keeping a list of all the state changes
2. waiting checks each entry and if the relevant state is found, all
   states up to and including then are removed.

This ensures that any waits will see all the state sets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
5257093268 tests/webrtc: factor out src pad property checking to a separate function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
2377f8b3f2 webrtcbin: initial support for sending and receiving simulcast streams
Input (sink pads) is the already-ssrc-muxed stream with the relevant rtp
sdes header extensions already applied:
  - mid
  - stream-id
  - repaired-stream-id

Output (src pads) have the pads separated into individual ssrc's as
that's what rtpbin gives us.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
75b23d646a tests/webrtc: test for enabled bundled fec/rtx
Doesn't actually check that any fec/rtx happens, just that the pipeline
is vaguely sane and doesn't error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
e18ee04cd2 tests/webrtc: also check valid mline for srcpad codec-preferences negotiation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
8a65fa40c7 webrtc/tests: print the correct media idx on error
Instead of the attribute index

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
b153ffdd56 webrtc/tests: give slightly better names to the dot file dumps
Don't use printf-specifiers with g_strconcat().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
cda81bdb1e webrtcbin: improve some debugging output
- Put human readable names into debug strings.
- Demote some frequent rtpbin signal logging
- Don't use GST_PTR_FORMAT in g_set_error()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
c02c8a85ce webrtcbin: silence spurious warning when creating answer transceiver
When creating a transceiver when creating an answer, the media kind of the
transceiver was never set correctly initially.  This would lead to a
GST_WARNING being produced about changin a transceiver's media kind.

Fix by retrieving the GstSDPMedia kind from the offer instead as the answer
GstSDPMedia has not been set as the answer caps have not been chosen yet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
246374c4e7 tests/webrtc: always use a unique SSRC for each stream
Will become more relevant with mid/rid->ssrc mappings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
9a758d78a9 webrtcbin: support using an a=mid value from the sink/transceiver caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
2e69886a02 ccconverter: ensure correct ordering of cea608 across output buffers
e.g. if a 60fps output is configured, we can only produce a single field
of cea608 that must alternate between field 1 and field 2.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2019>
2022-03-26 00:00:36 +00:00
Matthew Waters
6977119f99 ccconverter: ignore padding cea608 data even if marked as 'valid'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2019>
2022-03-26 00:00:36 +00:00
Mathieu Duponchelle
29de0e8e1d Revert "webrtcbin: fix msid line and allow customization"
This reverts commit 3cad3455377d5a22faa138d9df840257059776c8.

That commit was breaking the association between an audio and
a video track in the standard case.

In practice, to support carrying separate MediaStream, we are
going a way to map what MediaStreamTrack belong to what MediaStream,
but that will require some thinking about the API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2023>
2022-03-25 00:31:58 +01:00
Mathieu Duponchelle
06fec40f45 webrtcbin: fix msid line and allow customization
From https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-msid-16:

> Multiple media descriptions with the same value for msid-id and
> msid-appdata are not permitted.

Our previous implementation of simply using the CNAME as the msid
identifier and the name of the transceiver as the msid appdata was
misguided and incorrect, and created issues when bundling multiple
video streams together: the ontrack event was emitted with the same
streams for the two bundled medias, at least in Firefox.

Instead, use the transceiver name as the identifier, and expose
a msid-appdata property on transceivers to allow for further
customization by the application. When the property is not set,
msid-appdata can be left empty as it is specified as optional.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2003>
2022-03-24 16:43:29 +00:00
Branko Subasic
2689277a6b rtponviftimestamp: add support for using reference timestamps
Make it posible to configure the element to obtain the timestamps from
reference timestamp meta data instead of using the ntp-offset property,
or estimating its own offset. Currently the only time format supported
is "timestamp/x-unix", i.e. UTC time expressed in the unix time epoch.

In addition the custom event GstNtpOffset has been renamed to
GstOnvifTimestamp, to reflect that it is not necessarily used to convey
the ntp-offset. As a consequence we had to modify a couple of files in
the rtsp-server as well.

Fixes #984

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1683>
2022-03-11 08:39:50 +00:00
Matthew Waters
b7d0ddd1a4 webrtc: support renegotiating adding/removing RTX
We need to always add the RTX/RED/ULPFEC elements as rtpbin will only
call us once to request aux/fec senders/receivers.

We also need to regenerate the media section of the SDP instead of
blindly copying from the previous offer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1762>
2022-03-04 19:21:59 +11:00
Seungha Yang
04b8dfa391 d3d11: Add support for AYUV, AYUV64, and RGBA64_LE formats
Note that AYUV and AYUV64 formats will be used to expand format
support, especially some packed YUV formats (e.g., Y410, YUY2)
are common DXGI formats used for hardware decoder/encoder on Windows
but those formats cannot be used as a render target. We need to handle
them differently without pixel shader help, using compute shader
for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1699>
2022-02-16 18:41:05 +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
Seungha Yang
40213b5c75 av1parse: Use descriptive profile name instead of numeric
As per AV1 specification Annex A, AV1 profiles have explicit and
descriptive names for each seq_profile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1456>
2021-12-21 22:20:30 +09:00
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
Timo Wischer
214691b972 test: avtp: crf: Check for rounding errors
on average period calculation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1073>
2021-11-09 10:59:00 +00:00
Timo Wischer
5a25eb61b7 avtp: crf: Use double for average period calculation
to also support CRF intervals like every 1,333,333ns 64 events

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1073>
2021-11-09 10:59:00 +00:00
Timo Wischer
6a576938ac tests: avtp: crf: Test for timestamp_interval > 1
in case of CRF AVTPDUs with single CRF timestamp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>
2021-11-09 09:07:01 +01:00
Seungha Yang
aa1d97b0fb tests: h264parse: Add test for AUD insertion
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1296>
2021-11-04 16:32:54 +00:00
Mathieu Duponchelle
fa1805d531 cccombiner: stop attaching caption buffers when caption pad has gone EOS
When schedule is true (as is the case by default), we insert padding
when no caption data is present in the schedule queue, and previously
weren't checking whether the caption pad had gone EOS, leading to
infinite scheduling of padding after EOS on the caption pad.

Rectify that by adding a "drain" parameter to dequeue_caption()

In addition, update the captions_and_eos test to push valid cc_data
in: without this cccombiner was attaching padding buffers it had
generated itself, and with that patch would now stop attaching
said padding to the second buffer. By pushing valid, non-padding
cc_data we ensure a caption buffer is indeed attached to the first
and second video buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
2021-10-26 14:48:13 +00:00
Mats Lindestam
9b3f040d3c curlsftpsink: Add support for sha256 fingerprint
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1193>
2021-10-26 12:18:40 +00:00
Tim-Philipp Müller
e7c8b23b71 ofa: remove ofa audio fingerprinting plugin
I think the MusicIP database for this has been defunct for years,
so I can't imagine this plugin is particularly useful or still
used by anyone.

See https://musicbrainz.org/doc/Fingerprinting#PUID

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1153>
2021-10-17 10:17:00 +00:00
Víctor Manuel Jáquez Leal
1cfae69bc9 tests: va: Add VA buffer copy tests.
It should only work for raw buffers, but fails on dmabuf since it
should have a drm modifier.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1023>
2021-10-10 17:03:29 +00:00
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
Marc Leeman
a774dfb18f rtpmanagerbad: do not set iface on sink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/994>
2021-10-01 20:31:17 +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