Commit graph

182 commits

Author SHA1 Message Date
Johan Sternerup
1842ffc906 webrtc: Improve robustness of nice agent signal handlers
NiceAgent and it's associated thread is alive for as long as
GstWebRTCICE is alive so make sure any signal handlers connected to
NiceAgent do not access data that is deleted earlier.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2073>
2022-04-04 02:10:35 +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
Xavier Claessens
f270f9e974 Fix cross build with mingw32
At least on Ubuntu 20.04 the x86_64-w64-mingw32-gcc toolchain defaults
to WinXP. We require at least Vista for FILE_STANDARD_INFO.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2022>
2022-04-01 15:52:28 +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
041eee6c2e webrtc: produce stats for all relevant streams
Instead of only using the last ssrc that was pushed into a sink pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:41 +00:00
Matthew Waters
04de1a161f webrtc: avoid different versions of gnu-indent always wanting to change !!
Add some sneaky parenthesis to avoid always having to use git commit -n
or revert out hunk of the change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:41 +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
5741ee38e0 webrtc/datachannel: fix use-after-free in sctp state notification
g_signal_disconnect*() doesn't stop any existing callbacks from running
which means that if the notify::state callback is in progress in one
thread and the data channel object is finalize()ed in another thread,
then there could be a use-after-free trying lock the data channel
object.

We can't reasonably use a GWeakRef as we don't have a 'parent' object to
free the GWeakRef after the data channel is finalized.  This is also
complicated by the fact that the application can hold a reference to the
data channel object that would live beyond the lifetime of webrtcbin
itself.

We solve this by implementing a ghetto weak-ref solution internally with
a list of outstanding data channels.

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
699739c130 webrtcbin: support multiple received streams for a single mline
Each rtpbin exposed recv_src pad is now exposed as webrtcbin src_%u pad
now with no meaining applied to the value of %u.  Previously this used
to mean the mline in the SDP.  If this is is still required, then the
transceiver can be retrieved from the pad and the "mlineindex" property
from the transciever.  The "mid" is also retrievable from the
transceiver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
e28c45fd05 webrtc: explicitly error out in a couple of renegotiation cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
318a639e43 webrtc/transportstream: add debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
2aeca9ed84 webrtcbin: don't name src pads based on the mline specifically anymore
Naming based on the mline doesn't really work with e.g. simulcast
scenarios.

It is entirely possible to retrieve the transceiver and then the mline
from that if that is so required.

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
533d4937fe webrtcbin: add a specific find_transceiver_by_mid function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
79d58200c9 webrtcbin: explicitly use a variable for the rtp session idx
Slightly clearer in meaning.

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
fc28db57ae resindvd: silence unused-but-set warning
../ext/resindvd/gstpesfilter.c:117:11: error: variable 'STD_buffer_size_bound' set but not used [-Werror,-Wunused-but-set-variable]
  guint16 STD_buffer_size_bound;
          ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
2022-03-28 10:30:23 +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
Chun-wei Fan
b9f29bfc39 openexr: Specify modules when finding OpenEXR.
Specify modules to look for OpenEXR when CMake is used, as we may have
CMake config files instead of pkg-config files that result from building
OpenEXR, which may be built with CMake which is typically the case on Visual
Studio builds.

In this case, Meson does seem to find the 'OpenEXR' package with CMake
after trying pkg-config, but does not consider it enough without the
'modules:' argument.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2014>
2022-03-25 07:45:54 +00:00
Sangchul Lee
952c1194f3 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/2005>
2022-03-25 07:01:40 +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
Thibault Saunier
fe16900dff wpe: Mark first audio buffer as discont
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1492>
2022-03-24 00:01:20 +00:00
Thibault Saunier
a14e36fde4 wpe: Use about:blank as default URL to support only using load-bytes
WebKit is not going to render anything until a URI is set, leading to a
WPE posting a `WPE View did not render a buffer` error message. To avoid
requiring the user to know it if they only want to use
`wpesrc::load-bytes` we can just use `about:blank` as default and
everything will work as users would expect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1492>
2022-03-24 00:01:20 +00:00
Seungha Yang
454e8f58a8 aom: av1enc: Specify Temporal Unit alignment
Encoded bitstream consists of leading Temporal delimiter OBU
with frame, that's Temporal Unit alignment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1471>
2022-03-23 19:16:25 +00:00
Vivienne Watermeier
e6b187032b wpevideosrc: Add touch event support
Dispatches a list of active touch events to the wpe view on each
received TOUCH_FRAME event. Touch inputs currently only move the cursor,
since wpe doesn't seem to support clicking/scrolling or zooming with
touch input.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
2022-03-23 13:14:52 +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
Nirbheek Chauhan
8819350b74 openexr: Fix some warnings
```
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:46:24: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   46 |   virtual Int64 tellg ();
      |                        ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:47:32: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   47 |   virtual void seekg (Int64 pos);
      |                                ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:67:26: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   67 | Int64 MemIStream::tellg ()
      |                          ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:73:29: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   73 | MemIStream::seekg (Int64 pos)
      |                             ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1977>
2022-03-18 22:49:16 +00:00
Nirbheek Chauhan
253ee75a72 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/1864>
2022-03-18 08:16:46 +00:00
Matthew Waters
098ff9a453 ccconverter: drop data with a warning if scratch buffers overflow
Instead of asserting which could bring down the entire application.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1972>
2022-03-17 21:46:44 +11:00
Philippe Normand
3e3ba1772c wpe: Reintroduce persistent WebContext
A WebContext leak was introduced in MR
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252.
If we wanted one WebContext per WebView we should also unref the
WebKitWebContext when destroying the WebView.

This patch reintroduces the persistent WebContext, initially part of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1484.

Fixes #1084

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1933>
2022-03-16 09:07:21 +00:00
Mathieu Duponchelle
30d028317b webrtcbin: fix deadlock when setting up FEC encoder
We bind transceivers' fec_percentage property to the FEC encoder
percentage property, and with the binding bidirectional a deadlock
was introduced by the latest changes from !1762:

We take hold of the transceiver's object lock, then add the binding
and set the property to its initial value on the encoder, which causes
set_property to deadlock in the transceiver when the binding kicks in.

Changing the binding type to DEFAULT (source to target) is enough
to address the deadlock and still serves the original intent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1967>
2022-03-16 06:06:39 +00:00
Sangchul Lee
2f7c843f2b 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/1957>
2022-03-16 03:31:08 +00:00
Matthew Waters
ccd1b76625 webrtcbin: fix ulpfecenc passthrough pt
ulpfecenc uses a value of pt=255 for passthrough.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1075
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1914>
2022-03-10 16:20:03 +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
Sebastian Fricke
0b6bbce012 Remove the uninstalled term
Remove the symbolic link `gst-uninstalled` which points to `gst-env`.
The `uninstalled` is the old name and the project should stick to a
single name for the procedure.
Remove the term from all the files, exceptions are variables from
dependencies like `uninstalled_variables` from pkgconfig and
`meson-uninstalled`.
Adjust mentions of the script in the documentation and README.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Guillaume Desmottes
1f02f24828 gs: look for google_cloud_cpp_storage.pc
storage_client.pc was legacy and has been removed:
df6fa3611c (diff-bc35ad7c2fe631fd5578a06092412dba81c7ddd27bb25df7e17bb13771799afcL743)

No need to keep looking for storage_client.pc as a fallback as 1.25.0,
our minimum version, already ships google_cloud_cpp_storage.pc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1815>
2022-03-01 08:10:39 +00:00
Sanchayan Maity
7c9a315578 ldac: Set eqmid in caps
We set the eqmid in caps to be usable downstream by rtpldacpay for
knowing the frame count.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
2022-02-26 17:05:22 +05:30
Xavier Claessens
3d8372cc50 devenv: Add some missing GStreamer specific env variables
This should make "meson devenv" closer to what "gst-env.py" sets.

- GST_VALIDATE_SCENARIOS_PATH
- GST_VALIDATE_APPS_DIR
- GST_OMX_CONFIG_DIR
- GST_ENCODING_TARGET_PATH
- GST_PRESET_PATH
- GST_PLUGIN_SCANNER
- GST_PTP_HELPER
- _GI_OVERRIDES_PATH

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1768>
2022-02-25 20:35:26 +00:00
Jan Alexander Steffens (heftig)
e10bd02e1d fdkaacdec: Support arbitrary channel configs
Try to match the config to GStreamer positions. If something doesn't
fit, fall back to a set of unpositioned channels.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1561>
2022-02-25 18:20:52 +00:00
Jan Alexander Steffens (heftig)
d4b4ffc944 fdkaacdec: Use predefined channel layouts
This limits the decoder to the layouts predefined for the encoder
(including the MPEG standard layouts) but greatly simplifies the
implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1561>
2022-02-25 18:20:52 +00:00
Nicolas Dufresne
bab9041c4b Port plugins to gst_video_format_info_extrapolate_stride()
This reduces code duplication and simplify addition of new
pixel formats into related plugins.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
2022-02-20 22:32:55 +00:00
Jan Alexander Steffens (heftig)
10904e5580 wpe: Clean up build script
Use feature.require to check for gstgl and exit early if 'wpe' is
disabled (don't even check for wpe-webkit-1.1).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1668>
2022-02-20 14:34:12 +00:00
Martin Reboredo
717009f8f5 vulkanshaderspv: SPIRV based filter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1197>
2022-02-19 13:55:32 -03:00
Tim Mooney
97720dabe0 meson: check for libsocket and libnsl
If present, add '-lsocket' and '-lnsl' to network_deps.

ext/curl/meson.build: add network_deps to dependencies
gst/festival/meson.build: same
sys/shm/meson.build: same

Fixes linking issues on Illumos distros.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1525>
2022-02-17 18:44:49 +00:00
Jan Alexander Steffens (heftig)
acd0300485 openaptx: Support libfreeaptx
[libfreeaptx][1] is a fork of libopenapt 0.2.0, used by pipewire.

[1]: https://github.com/iamthehorker/libfreeaptx

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1642
Closes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1589
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1667>
2022-02-15 08:18:44 +00:00
Sangchul Lee
dcff37722d webrtcice: Fix memory leaks in gst_webrtc_ice_add_candidate()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1646>
2022-02-09 09:00:25 +00:00
Stéphane Cerveau
0600acd715 dashsink: doc cleanup
Remove max-files mention in the command line test
Fix some typos
Use mpegtsdemux instead of tsdemux in the pipeline description

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1624>
2022-02-02 10:21:08 +01:00
Jan Alexander Steffens (heftig)
16dc8f8442 wpe: Support wpe-webkit-1.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1522>
2022-01-31 08:31:34 +00:00
Philippe Normand
8e4cce6cd3 wpe: Install WebExtension in pkglibdir
The uninstalled WebExtension takes precedence over the installed one, so that
audio support works in local developer builds as well.

Fixes #975

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1602>
2022-01-31 00:54:10 +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
Robert Mader
e7c9960783 waylandsink: Ensure correct mapping of area_surface
If the `area_surface` got unmapped when changing to the `READY` or
`NULL` state, we currently don't remap it when playback resumes and
`wp_viewporter` is supported. Without `wp_viewporter` we do remap
it, but rather unintentionally and also when not wanted.

On Weston this has not been a big problem as it so far wrongly maps
subsurfaces of unmapped surfaces anyway - i.e. only the black
background was missing on resume. On other compositors and future
Weston this prevents the `video_surface` to get remapped.

Shuffle things around to ensure `area_surface` is mapped in the
right situations and do some minor cleanup.

See also https://gitlab.freedesktop.org/wayland/weston/-/issues/426

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1483>
2022-01-17 13:17:57 +00:00
Robert Mader
f0b04f1ef1 waylandsink: Use wl_surface_damage_buffer() instead of wl_surface_damage()
The later, doing damage in surface coordinates instead of buffer
coordinates, has been deprecated. The reason for that is that it
is more prone to bugs, both on the client and the compositor side,
especially when paired with buffer scale, `wp_viewporter` or
buffer transforms.

Unfortunately, on Weston this risks running into
https://gitlab.freedesktop.org/wayland/weston/-/issues/446
(which causes trouble for several other projects as well). However,
that bug only affects cases where we run in sync mode, i.e. only
during resizes. In practise I haven't been able to observe the
issue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
1249362f96 waylandsink: Use G_MAXINT32 for surface damage
Each time we call `wl_surface_damage()` we want to do full surface
damage. Like Mesa, just use `G_MAXINT32` to ensure we always do
full damage, reducing the need to track the right dimensions.

`window->video_rectangle` is now unused, but we keep it around for
now as we may need it again in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
3bbd091bb4 waylandsink: Only call wl_surface_damage() when buffer content changed
From the spec:
> This request is used to describe the regions where the pending
> buffer is different from the current surface contents

We currently also call `wl_surface_damage()` on surfaces without
new or still compositor-hold buffers, e.g. when resizing the window.
In that case we call it on `area_surface_wrapper`, even though it
gets resized via `wp_viewport_set_destination()`, in which case
the compositor is in charge of repainting the area on screen.

Doing so is currently not forbidden by the spec, however it might
be in the future, see
https://gitlab.freedesktop.org/wayland/wayland/-/issues/267

Thus lets stay close to the spec and only call `wl_surface_damage()`
when we just attached a buffer.

Right now this prevents runtime assertions in Mutter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
b03c7edfcf waylandsink: Simplify input region handling
We only need to unset the input region for the area surface when
we don't have our own toplevel surface. By default, the input region
covers the whole surface, thus no need to change it on resize.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
1e2bc68171 waylandsink: Use G_MAXINT32 for opaque regions
`gst_wl_window_set_opaque` does not get called on window resizes,
potentially leaving opaque regions too small.
According to the spec opaque regions can be bigger than the surface
size - parts that fall outside of the surface will get ignored.
Thus we can can simply use `G_MAXINT32` and be sure that the whole
surfaces will always be covered.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +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
Mathieu Duponchelle
d8c8737e71 cccombiner: fix s334-1a scheduling
The previous code was mistakenly trying to compute a cc_type out
of the first byte in the byte triplet, whereas it is to be interpreted
as:

> Bit b7 of the LINE value is the field number (0 for field 2; 1 for field 1).
> Bits b6 and b5 are 0. Bits b4-b0 form a 5-bit unsigned integer which
> represents the offset

The same mistake was made when creating padding packets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1496>
2022-01-12 14:34:22 +00:00
Mathieu Duponchelle
6861ea8fe1 cccombiner: merge buffers for both fields with caption type s334-1a
Other elements such as line21encoder expect both fields to be present
in the same meta, not one meta per field.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1496>
2022-01-12 14:34:22 +00:00
Nirbheek Chauhan
1be6d6ccf5 meson: Add explicit check: kwarg to all run_command() calls
This is required since Meson 0.61.0, and causes a warning to be
emitted otherwise:

2c079d855e
https://github.com/mesonbuild/meson/issues/9300

This exposed a bunch of places where we had broken run_command()
calls, unnecessary run_command() calls, and places where check: true
should be used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
2022-01-09 18:12:47 +05:30
Rafał Dzięgiel
8889b6351d assrender: Support RFC8081 mime types
Old "application/*" are now as per RFC8081 deprecated in favor of
new "font/*" mime types. Some new encoders are already using the
updated mime types. We need to also add them to the support list
in order for assrender to correctly identify them as fonts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1481>
2022-01-03 06:42:23 +00:00
Rafał Dzięgiel
a2719d79ff assrender: Handle ".ttc" attachment extension
TTC stands for "TrueType Collection" file. We can pass it
into libass as any other attachment. Add it to the supported
extensions list, so the fonts it contains will be used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1481>
2022-01-03 06:42:23 +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
Seungha Yang
796007f75d av1enc: Update for newly designed AV1 profile signalling
Accept named AV1 profiles (i.e., main, high, and professional)
as well

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1456>
2021-12-21 22:20:34 +09:00
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
Thibault Saunier
d82efb47aa pitch: Specify layout as required for negotiation
There are cases where it might negotiate 'non-interleaved' while it
is wrong.

```
gst-launch-1.0 audiotestsrc !  "audio/x-raw, format=(string)F32LE, layout=(string)non-interleaved" ! audioconvert ! audioresample ! pitch tempo=1.2 ! audioconvert ! "audio/x-raw,format=S16LE" ! fakesink

Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
(gst-launch-1.0:3029628): GStreamer-Audio-CRITICAL **: 11:42:22.477: gst_audio_buffer_map: assertion '(!meta && info->layout == GST_AUDIO_LAYOUT_INTERLEAVED) || (meta && info->layout == meta->info.layout)' failed
ERROR: from element /GstPipeline:pipeline0/GstAudioConvert:audioconvert1: The stream is in the wrong format.
Additional debug info:
../subprojects/gst-plugins-base/gst/audioconvert/gstaudioconvert.c(876): gst_audio_convert_transform (): /GstPipeline:pipeline0/GstAudioConvert:audioconvert1:
failed to map input buffer
ERROR: pipeline doesn't want to preroll.
ERROR: from element /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0: Internal data stream error.
Setting pipeline to NULL ...
Additional debug info:
../subprojects/gstreamer/libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0:
streaming stopped, reason error (-5)
ERROR: pipeline doesn't want to preroll.
Freeing pipeline ...
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1441>
2021-12-11 19:09:09 -03:00
Philippe Normand
86719e25a4 wpevideosrc: Use basesrc event vfunc
Allows for basic default handling from the base class.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1422>
2021-12-07 11:43:26 +00:00
Tim-Philipp Müller
26169cee0e teletextdec: fix minor string leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1416>
2021-12-06 13:07:37 +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
Seungha Yang
2a17618dcc openjpegenc: Fix build warning
Compiling C object subprojects/gst-plugins-bad/ext/openjpeg/gstopenjpeg.dll.p/gstopenjpegenc.c.obj
../subprojects/gst-plugins-bad/ext/openjpeg/gstopenjpegenc.c(416):
  warning C4133: '=': incompatible types - from 'GstFlowReturn (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)' to
  'gboolean (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)'

../subprojects/gst-plugins-bad/ext/openjpeg/gstopenjpegenc.c(418):
  warning C4133: '=': incompatible types - from 'GstFlowReturn (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)' to
  'gboolean (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1378>
2021-11-24 13:11:23 +00:00
Guillaume Desmottes
d67a63a298 gssink: add metadata property
This property can be used to set metadata on the storage object.

Similar API has been added to the S3 sink already, see
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/613

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1377>
2021-11-23 16:00:53 +01:00
Philippe Normand
a6fd767025 wpevideosrc: Fix frame stuttering in GL rendering path
Make sure the EGLImage we're rendering to the GL memory stays alive long enough,
until the the GL memory has been destroyed.

This change fixes tearing and black flashes artefacts that were happening
because the EGLImage was sometimes destroyed before the sink actually rendered
the associated texture.

Fixes #889

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1354>
2021-11-16 21:55:41 +00:00
Philippe Normand
053dd564a1 wpevideosrc: Run through gst-indent
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1354>
2021-11-16 21:55:41 +00:00
Tim-Philipp Müller
972615cf22 docs: fix unnecessary ampersand, < and > escaping in code blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1340>
2021-11-12 11:39:19 +00:00
Timo Wischer
8e7ce64a6e avtp: crf: Process also local CRF streams
Without this patch locally generated CRF streams will be ignored.
Therefore the same network interface could not be CRF talker and
CRF listener.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1074>
2021-11-10 16:53:04 +00:00
Timo Wischer
36006c61e9 avtpsrc: Use correct size for provided buffers
Without this patch the following pipeline would send packets containing
garbage in the data section.
$ gst-launch-1.0 avtpsrc ! avtpsink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1077>
2021-11-09 16:59:10 +00:00
Timo Wischer
de95d3a1c4 avtp: crfsync: Warn when CRF package not yet received
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1075>
2021-11-09 15:36:25 +01:00
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
5a9e9895ab avtp: crf: Properly handling one timestamp per PDU
The average_period should always represent the time between two
events. The specification defines the event time as the time
between audio samples, video frame sync, video line sync, etc.
In case of one timestamp per PDU the timestamp_interval identifies
the amount of events between the timestamp of one PDU and the
timestamp of the next PDU.
As described in IEEE 1722-2016 chapter
"10.4.12 timestamp_interval field" timestamp_interval shall be
nonzero.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>
2021-11-09 09:07:01 +01:00
Martin Reboredo
2546cef4be aom: Set fixed_qp_offsets to a deactivated value
aom only uses fixed_qp_offsets with the
Constant Quality (Q) Rate Control mode,
previously this was locking any usage
with another Rate Control mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1198>
2021-11-08 16:42:17 +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
Tim-Philipp Müller
1f560af76b dtls: don't use deprecated g_binding_get_source() with newer GLib versions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1279>
2021-10-30 00:52:42 +01:00
Heiko Becker
b83e85ab67 neon: Allow building against neon 0.32.x
No API/ABI changes: https://github.com/notroj/neon/blob/0.32.0/NEWS#L3

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1267>
2021-10-29 00:14:53 +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