Commit graph

61 commits

Author SHA1 Message Date
Seungha Yang 37fdaaf8ff proxysink: Make sure stream-start and caps events are forwarded
There might be a sequence of event and buffer flow:
- Got stream-start/caps/segment events
- Got flush events
- And then buffers with a new segment event

In the above case, stream-start and caps event might not be reached to
peer proxysrc if peer proxysrc is not ready to receive them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1552>
2022-07-07 05:42:21 +09:00
Tim-Philipp Müller c895cdbec8 tests: skip unit tests for dependency-less elements that have been disabled
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1136

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2660>
2022-06-27 07:05:00 +00:00
Seungha Yang 72975fbd6d h264parser: Add an API for AVCDecoderConfigurationRecord parsing
Add a method for AVC configuration date parsing

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2449>
2022-06-15 19:58:59 +00:00
Philippe Normand c287711418 webrtcbin: Add a prepare-data-channel GObject signal
This new signal allows data-channel consumers to configure signal handlers on a
newly created data-channel, before any data or state change has been notified.

The webrtcin unit-tests were refactored to make use of this new signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2427>
2022-06-07 11:29:33 +00:00
Olivier Crête 9fe2e1c5eb webrtcbin: Reject answers that don't contain the same number of m-line as offer
Otherwise, it segfaults later. Also add test to validate this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2526>
2022-06-03 20:28:19 +00:00
Brad Hards 804a6054bb h264parse: add unit test for Precision Time Stamp in SEI messages
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1458>
2022-06-03 08:29:05 +00:00
U. Artie Eoff becabd36da tests: va: add simple vacompositor test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2481>
2022-05-27 09:42:36 +00:00
Sherrill Lin f335b40ae8 webrtcstats: Update unit test for outbound rtp stats
"remote-id" is not guaranteed to present after commit 1deb034e3d.
Thus, we should not fail the test if "remote-id" is not found.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Sherrill Lin 3e7fb83393 webrtcstats: Improve selected candidate pair stats by adding ICE candidate info
The implementation follows w3.org specs:
* https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*
* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*

Corresponding unit tests are also added.

Rebased and updated from
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462

Fixes #1207

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Matthew Waters be2dfd0c36 webrtcbin: reuese the same fec/rtx/red payload types for the same media payload
WHen bundling, if multiple medias are used with the same media payload, then
each of the fec/rtx/red additions would add a distinct payload.  This could
very easily overflow the available payload space.

Instead, track the relationship between the media payload value and
the relevant fec/rtx/red payload values and reuse them whenever
necessary, even when bundling.

e.g.

...
a=group:BUNDLE video0 video1
m=video 9 UDP/SAVPF 96 97
a=mid:video0
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...
m=video 9 UDP/SAVPF 96 97
a=mid:video1
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2474>
2022-05-24 10:21:11 +00:00
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
Seungha Yang be84fc23ca h265parser: Add a new NAL parsing API to handle malformed packets
Add gst_h265_parser_identify_and_split_nalu_hevc() method to
handle a case where packetized stream contains start-code prefix.
This new method behaves similar to exisiting gst_h265_parser_identify_nalu_hevc()
but it will scan start-code prefix to split given data into
NAL units.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2394>
2022-05-10 03:58:51 +09:00
Mengkejiergeli Ba efdd63d875 tests: Skip test if srtp element not built
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2383>
2022-05-06 09:13:31 +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
He Junyan d824698561 test: Add test cases for the H265 bitwriter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1831>
2022-04-22 07:35:17 +00:00
Xavier Claessens b004464ac6 Remove glib and gobject dependencies everywhere
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.

While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
2022-04-01 16:32:17 +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
Thibault Saunier 25819c41fb navigation: Add support for key Modifiers in all relevant events
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2010>
2022-03-25 15:16:03 +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
Vivienne Watermeier 6c2f6c3bd4 all: Use new navigation interface and API
Use and implement the new navigation interface in all relevant sink elements,
and use API functions everywhere instead of directy accessing the event structure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
2022-03-23 13:14:52 +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
Philippe Normand 21f7889187 gstplay: tests: Keep track of errors/warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1869>
2022-03-07 11:03:41 +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
He Junyan 1defc9ce6b test: Add test cases for the H264 bitwriter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:49 +01:00
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
Víctor Manuel Jáquez Leal 69c4e317d8 tests: h265parser: Add test for multiple compatibility profiles.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1440>
2021-12-16 17:08:30 +01:00
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