Commit graph

72 commits

Author SHA1 Message Date
He Junyan
2408ca2f18 h265bitwriter: Correct the all API to byte aligned.
In fact, all the h265 bit writer have byte aligned output. So we
change the API from bit size in unit to byte size, which is easy
to use.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3193>
2022-10-27 12:15:43 +00:00
He Junyan
c294ba82e6 h264bitwriter: Correct the all API to byte aligned.
In fact, all the h264 bit writer have byte aligned output except
the slice header. So we change the API from bit size in unit to
byte size, which is easy to use. For slice header, we add a extra
"trail_bits_num" to return the unaligned bits number.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3193>
2022-10-27 12:15:43 +00:00
Tim-Philipp Müller
e703374ff8 fdkaac: add minimal unit test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:05 +00:00
Sangchul Lee
0f4cf19fb9 tests/webrtc: Add test for 'add-turn-server' action signal
It just checks return value of the action signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3131>
2022-10-11 10:23:00 +00:00
Johan Sternerup
212c09a70e webrtc: return error when sending on non-open datachannel
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.

Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
2022-10-05 11:08:30 +00:00
Stéphane Cerveau
fb09c028e3 h265parse: fix typo in member of GstH265SPS
Rename sps_extnsion_params to sps_extension_params

Fix comment about vui_parameters_present_flag

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3072>
2022-10-04 10:01:12 +00:00
Devin Anderson
31831eb47e voamrwbenc: Fix truncation of audio data at end-of-stream when audio data
doesn't align on 20 millisecond frame size.

The AMR-WB codec imposes a fixed 20 millisecond frame size.  In its current
form, the `voamrwbenc` plugin deals with this limitation by discarding any
audio at the end of the stream that falls short of 20 milliseconds.  This patch
keeps the audio data, and appends silence to the end to preserve frame size
alignment.

The patch also adds tests to check for the updated behavior.  I noticed that
tests weren't being built, so I changed the build to allow for building the
tests when the `tests` and `voamrwbenc` options are set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3027>
2022-09-16 00:14:58 +00:00
Tim-Philipp Müller
2ac5d687e1 tests: add a few more orc tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3029>
2022-09-15 12:14:56 +01:00
Olivier Crête
4b3b234f72 webrtcbin: Allow locked mlines with no caps, as the last ones
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
2022-09-02 11:52:58 +02:00
Olivier Crête
0930c467d4 webrtcbin: Reject creating an offer if a locked mline has no caps
This avoids getting in a bunch of corner cases. We'd have to insert
a "rejected" line from the start as a place-holder to get around this,
but the rest of the code just becomes more complicated, so just
disallow it for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
2022-09-02 11:52:58 +02:00
Olivier Crête
3503599e0a webrtcbin: Store pending mid to make create-offer idempotent
If the mid is not stored in the transceiver, but it is stored in
last_offer, then a further create-offer call will just ignore that
transceiver.

Also include unit test for ensure it doesn't regress.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
2022-09-02 11:52:58 +02:00
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