Commit graph

7209 commits

Author SHA1 Message Date
Philippe Normand
a0b37e9d1a wpe: Bump wpebackend-fdo version requirement to 1.8
Debian bullseye has this version already, and this allows us to get rid of many
ifdefs. The mouse scroll handling is actually functional now as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2278>
2021-05-23 17:18:20 +00:00
Tim-Philipp Müller
0151276d7f Use new gst_buffer_new_copy()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2279>
2021-05-23 17:20:16 +01:00
Thibault Saunier
818db8f0b3 wpe: Bump WPE dependency to 2.28
The new audio feature depends on WPE 2.28 so we should just bump our
requirement to that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2264>
2021-05-19 18:50:29 -04:00
Thibault Saunier
c98fe5b7f9 wpe: Update doc cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
2021-05-19 13:41:16 +00:00
Thibault Saunier
9415106b02 wpe: Properly respect LIBGL_ALWAYS_SOFTWARE
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
2021-05-19 13:41:15 +00:00
Thibault Saunier
4dbfae0105 wpe: Relay messages from WPE internal pipelines
It is based on a tracer as it allows us to very easily get
every message that are posted on any bus inside the process.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
2021-05-19 13:41:15 +00:00
Thibault Saunier
a92d4373ad wpe: Base wpe audio implementation on a web extension
This makes the implementation simpler and enable us to map
webviews and audio stream much more easily

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
2021-05-19 13:41:15 +00:00
Philippe Normand
81ced7932f wpe: Enable WebAudio
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
2021-05-19 13:41:15 +00:00
Philippe Normand
f4bc5c6c65 wpe: Implement audio support
The wpesrc bin now exposes "sometimes" audio src pads, one for every PCM audio
stream created by WPEWebKit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
2021-05-19 13:41:15 +00:00
Thibault Saunier
cb4f6c877e wpe: Move wpesrc to wpevideosrc and add a wrapper bin wpesrc
Currently the bin contains a single element but we are going
to implement audio support and expose extra pads for audio

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
2021-05-19 13:41:15 +00:00
Doug Nazar
be1c154f33 sctp: Ensure pad is still a child of element before removal
During pipeline shutdown there are several competing paths to remove
pads. Avoids tests failing due to:

Unexpected critical/warning: Padname '':sink_1 does not belong to element sctpenc1 when removing

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2256>
2021-05-19 05:49:48 +00:00
Doug Nazar
5663db236f sctp: Fix race of pad removal during reset/stop
Both reset & stop remove existing pads. Can result in warning from
gst_element_remove_pad().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2256>
2021-05-19 05:49:48 +00:00
Doug Nazar
4fcfd5b7f9 webrtcbin: Fix race bringing up sctp data channel
Notifying before pads are linked can cause the stream to fail to start.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2256>
2021-05-19 05:49:48 +00:00
Matthew Waters
a836bd4766 webrtcbin: advertise harder the rtcp-mux-only requirement
And ignore rtcp ICE candidates

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2239>
2021-05-19 04:42:56 +00:00
Sid Sethupathi
abe7e724ed webrtcbin: update default jb latency docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2242>
2021-05-19 03:53:55 +00:00
Doug Nazar
20ca07d174 dtls: Let sender know when we are flushing
Prevents endless loop during shutdown where we end up sending 0 bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2229>
2021-05-19 03:21:58 +00:00
Doug Nazar
8b8428aec2 dtls: Add ability to set custom GstFlowReturn on callback error
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2229>
2021-05-19 03:21:58 +00:00
Olivier Crête
3bdf1e691e webrtc: Remove reundundant context object in transportsendbin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2260>
2021-05-18 20:26:38 -04:00
Olivier Crête
51821644ba webrtc: Wait until ICE is connected to start DTLS handshake process
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2260>
2021-05-18 20:18:28 -04:00
Olivier Crête
b6965e9906 webrtcbin: Remove pad probe on nicesink
This pad probe is no longer necessary, libnice now drops
all buffers before the stream is connected. This pad problem
also caused deadlocks in some situations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2260>
2021-05-18 19:08:48 -04:00
Olivier Crête
28bd479ea2 kate: Initialize debug categories
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2258>
2021-05-18 13:22:49 +00:00
Olivier Crête
761206291b openh264: Don't use GOnce for ABI check
It turns out the value used for g_once_* APIs can't be
zero. And this is a very cheap check, so let's just do it every time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2240>
2021-05-13 21:40:02 +00:00
Olivier Crête
f34be8a3bd webrtcbin: Intersect answer with codec prefs & capabilities
In case the local capabilities changed since the last negotiaton,
we need to re-intersect and see if the result would be different.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
2021-05-13 15:05:00 -04:00
Olivier Crête
3065ac49fb webrtcbin: Ignore current caps for codec negotiation
On the sink pad, we want the caps of the current stream, those
are the "received_caps" field. If we haven't received caps yet, then
we only care about the caps that the next element can accept, that is
the caps from the peer pad (and the preferences). Otherwise, we prevent
re-negotiation to a better codec when possible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
2021-05-13 15:05:00 -04:00
Olivier Crête
4bb94c6970 webrtcbin: Remove dead code
The function is only called to create an offer, so no
need to pass the offer parameter and then check it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
2021-05-13 15:05:00 -04:00
Olivier Crête
f6345b4b03 webrtcbin: Refactor codec preference retrieval
Now intersect against pads on both sides if they are available.
If the intersection fails, we now just reject the creation of the offer
or answer as it means that the codec_preferences are too restrictive or
that the caps on both sides the webrtcbin are not compatible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
2021-05-13 15:05:00 -04:00
Olivier Crête
48f4498801 webrtcbin: Intersect codec preferences with caps from pads
When creating an offer or an answer, also take into account
the caps on the pads as well as the codec preferences when both are set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
2021-05-13 15:05:00 -04:00
Olivier Crête
70befc0b21 webrtcbin: Implement caps queries on sinkpad based on codec preferences
Also includes a unit test.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
2021-05-13 15:05:00 -04:00
Olivier Crête
dc6655542d webrtcbin: Hold transceiver lock when accessing codec_preferences
This is required to allow the applications to modify the preferences.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
2021-05-13 15:05:00 -04:00
Johan Sternerup
4d514abfd6 webrtcbin: Fix deadlock when receiving new sctp stream
When receiving an sctp message for a stream that not yet has an
sctpdec pad associated with it means we end up in
_on_sctpdec_pad_added. At this point we're holding the sctpassocation
lock. Then it's not possible to take the pc_lock because then code
executing under the pc_lock (which means anything in the webrtc
thread) may not take the sctpassociation lock. For example, running
the data channel close procedure from the webrtc thread means we
eventually end up sending a SCTP_RESET_STREAMS packet which needs to
grab the sctpassociation lock.

This means _on_sctpdec_pad_added simply cannot take the pc_lock and
also it is not possible to postpone the channel creation as we need to
link the pads right there. The solution is to introduce a more
granular dc_lock that protects only the things that needs to be done
to create the datachannel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186>
2021-05-12 03:02:27 +00:00
Johan Sternerup
8dbdfad914 webrtcbin: Support closing of data channels
Support for closing WebRTC data channels as described in RFC
8831 (section 6.7) now fully supported. This means that we can now
reuse data channels that have been closed properly. Previously, an
application that created a lot of short-lived on-demand data channels
would quickly exhaust resources held by lingering non-closed data
channels.

We now use a one-to-one style socket interface to SCTP just like the
Google implementation (i.e. SOCK_STREAM instead of SOCK_SEQPACKET, see
RFC 6458). For some reason the socket interface to use was made
optional through a property "use-sock-stream" even though code wasn't
written to handle the SOCK_SEQPACKET style. Specifically the
SCTP_RESET_STREAMS command wouldn't work without passing the correct
assocation id. Changing the default interface to use from
SOCK_SEQPACKET to SOCK_STREAM now means we don't have to bother about
the association id as there is only one association per socket. For
the SCTP_RESET_STREAMS command we set it to SCTP_ALL_ASSOC just to
match the Google implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186>
2021-05-12 03:02:27 +00:00
Daniel Stone
4eeaa8f170 openaptx: Fix to v0.2.0 due to license change
openaptx has recently changed its license to explicitly exclude
'Freedesktop projects' from using it, which would include GStreamer, as
well as shifting to base terms of GPLv3:
    811bc18586

This unilateral license change is legally dubious in many ways.

The original work came from ffmpeg under the LGPL v2.1, to which third
parties may not add additional restrictions (per sections 2 and 7 of the
LGPL v2.1), so LGPLv2.1 + may-not-use restrictions are not permissible
without the explicit consent of the original copyright holder.

The upgrade to LGPL v3.0 without explicit consent from the original
copyright holder is in itself permissible through the upgrade terms of
the LGPL, however the additional restrictions imposed again conflict
with sections 7 and 10 of the GPLv3 (as the base of the LGPLv3, with
those sections not being invalidated by the additional LGPLv3 text).

Though it does not impact the legal validity of the redeclaration of
licensing, the claims that freedesktop.org has violated the terms of the
openaptx license in the past are false; the work was contributed to the
PulseAudio project with an explicit open license, with the original
contributor later attempting to revoke permission for its use, despite
the explicit terms of the license giving no ability to do so as they
lack a change-of-heart provision.

The claims that Collabora violated the license are even more baseless;
they are based on an assertion that when I (acting on behalf of
freedesktop.org rather than Collabora, in my own unpaid time) banned
users from freedesktop.org's GitLab instance due to sustained violations
of the Code of Conduct users agree to when creating an account on that
platform, this somehow constituted a violation of the license. Even if
Collabora were somehow involved in this - which they were not at all -
there is no requirement under open licenses that users be given
unlimited access under all terms to any platform on the internet. Such
terms would mean that open development could only be conducted on
completely unmoderated platforms, which does not stand up to any
scrutiny.

Regardless of the declared license having no legal validity, the LGPL's
explicit provision in both v2.1 and v3.0 for such additional
restrictions to be stripped, and the low likelihood of it ever being
used together with GStreamer as its licensing terms would not be
acceptable to any distribution, enforcing a version check seems like the
safest way to ensure complete legal clarity, not put users or
downstreams in any jeopardy, and comply with the author's stated wishes
for v0.2.1 and above to not be used by GStreamer.

Signed-off-by: Daniel Stone <daniel@fooishbar.org>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2235>
2021-05-11 14:53:56 +00:00
Jan Alexander Steffens (heftig)
c9a04ca979 wpe: Properly free property fields
The set location (in two places) and loaded bytes were not freed when
the element is destroyed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2222>
2021-05-07 16:01:49 +00:00
Jan Alexander Steffens (heftig)
950d5eedf9 wpe: Properly lock property fields
Use the object lock for the following fields:
 - `bytes`: Written by the `load-bytes` signal unless running; consumed
   on start.
 - `draw_background`: Read and written by the `draw-background`
   property.
 - `location`: Read and written by the `location` property and the URI
   handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2222>
2021-05-07 16:01:49 +00:00
Matthew Waters
a78c907597 webrtc: only add nack pli by default if kind is video
Sending/receiving PLI's (Picture Loss Indication) for non-video doesn't
really make sense.  This also matches what the browsers do.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2220>
2021-05-06 12:19:51 +00:00
Matthew Waters
1470660976 webrtc: move webrtc_kind_from_caps() to utils
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2220>
2021-05-06 12:19:51 +00:00
Sebastian Dröge
da0e6b0afc hlssink(2): Don't write deprecated EXT-X-ALLOW-CACHE metadata
It's deprecated since quite a few versions and various validators
complain about it. Instead of the in-manifest metadata this should be
handled by the normal HTTP caching headers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2221>
2021-05-06 10:46:15 +03:00
François Laignel
ad3d7d34cc Use gst_element_request_pad_simple...
Instead of the deprecated gst_element_get_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2180>
2021-05-05 06:17:14 +00:00
Antonio Rojas
51e96fd2c3 Fix build with OpenEXR 3
Add a header that is no longer transitively included

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2195>
2021-05-04 14:58:49 +00:00
Matthew Waters
5835f2aa8d webrtc: advertise support for transport-cc rtcp-fb by default
Still requires explicit enabling by the application through the header
extension on all the relevant payloaders.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 22:01:54 +10:00
Matthew Waters
1ab58736df webrtc/stats: provide the twcc stats when available
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 22:01:54 +10:00
Timo Wischer
2ef4639fe0 avtp: crf: Remove superfluous sink_event variable
This variable was introduced by commit 12ad2a4bcd ("avtp: Introduce
the CRF Sync Element") but it was never used:
$ git log -G "sink_event" -- ext/avtp

Signed-off-by: Timo Wischer <timo.wischer@de.bosch.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2201>
2021-04-28 08:59:14 +00:00
Aaron Boxer
f71eb29497 onnx: add plugin to apply ONNX neural network models to video
This MR provides a transform element that leverage ONNX runtime
to run AI inference on a broad range of neural network toolkits, running
on either CPU or GPU. ONNX supports 16 different providers at the
moment, so with ONNX we immediately get support for Nvidia, AMD, Xilinx
and many others.

For the first release, this plugin adds a gstonnxobjectdetector element to
detect objects in video frames. Meta data generated by the model is
attached to the video buffer as a custom GstObjectDetectorMeta meta.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1997>
2021-04-27 13:05:21 +00:00
Nazar Mokrynskyi
fe190fb5eb webrtcbin: downgrade "dropping ICE candidates from SDP" from warning to debug level
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2187>
2021-04-23 00:15:26 +00:00
Olivier Crête
c690be3e03 webrtcbin: Attach rtpbin even for data channels
This is required because the same transport may later be used for RTP.
In which case the RTCP needs to flow bi-directionnally already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2172>
2021-04-22 22:15:31 +00:00
Frederich Munch
6f2c010360 Fix missing unref of nice-agent causing sockets to never close.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1960>
2021-04-22 21:14:49 +00:00
Doug Nazar
4e29ba9fce webrtc: Fix sctp task's return type.
GstWebRTCBinFunc expects a GstStructure* return type.

Fixes segfault on PowerPC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2188>
2021-04-22 16:14:41 -04:00
Olivier Crête
813a320c06 webrtcbin: Filter caps isn't fixed
Fix an assertion because the filter paramter passed to
gst_caps_is_equal_fixed() wasn't fixed. So use the regular
gst_caps_is_equal() instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2175>
2021-04-19 19:06:50 -04:00
Philippe Normand
8b1051cdea webrtcdsp: Propagate VAD to audio level meta
Whenever the voice activity changed on the stream, update or create an
AudioLevelMeta and associate it to the corresponding buffer.

Fixes #1073

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2170>
2021-04-19 15:51:32 +00:00
Sebastian Dröge
c2635c154d cccombiner: Use correct enum when registering the max-scheduled property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2171>
2021-04-19 13:51:57 +03:00