Commit graph

167 commits

Author SHA1 Message Date
Sebastian Dröge
a54eddad3a webrtcbin: Reject caps that are not valid for creating an SDP media.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2689>
2022-06-30 09:28:27 +00:00
Tim-Philipp Müller
afc94046ba dv, opusparse: fix duplicate symbols in static build
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1262

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2670>
2022-06-28 01:29:06 +01:00
Robert Mader
6aa0b0cae2 gstwaylandsink: Add rotate-method property
Similar to and inspired by glimagesink and gtkglsink.

Using the Wayland buffer transform API allows to offload
rotate operations to the Wayland compositor. This can have
several advantages:
 - The Wayland compositor may be able to use hardware plane
   capabilities to do the rotation.
 - In case of pre-rotated content on rotated outputs the
   rotations may equal out, potentially allowing the
   compositor to use hardware planes even if they don't
   support rotate operations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2543>
2022-06-20 18:30:56 +00:00
Mathieu Duponchelle
f10e2eb88f cccombiner: expose output-padding property
When schedule=true and output-padding=false, cccombiner will not
inject padding in the output closed caption meta stream.

The property has no effect when schedule=false.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1621>
2022-06-17 14:11:46 +00:00
Olivier Crête
c4971a456e webrtcbin: Limit sink query to sink pads
This allows the reception of streams that don't exactly match
the codec preferences. In particular, the ssrc in the codec preferences
is local sender SSRC, the other side is expected to send a different SSRC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2615>
2022-06-17 08:08:43 +00:00
Stéphane Cerveau
19972b8153 srtsrc: add "keep-listening" property to avoid EOS on disconnect
The property 'keep-listening' avoids EOS
when the remote client disconnects.

It can be useful to a keep a pipeline alive
when the srt connection drops remotely.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/967>
2022-06-15 20:35:14 +00:00
Stéphane Cerveau
eb1f21b484 srtsrc: remove dead code
Remove code useless since
132e3a1af9

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/967>
2022-06-15 20:35:14 +00:00
Matthew Waters
9df7a21ec9 vulkan: add vulkan overlay compositor element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2470>
2022-06-14 03:34:06 +00:00
Matthew Waters
81e601ccaa vulkan: move element register definition to relevant element headers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2470>
2022-06-14 03:34:05 +00:00
Tim-Philipp Müller
9d9e59622f Bump GLib requirement to >= 2.62
Can't require 2.64 yet because of
https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
2022-06-10 06:01:41 +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
Philippe Normand
779ca38229 webrtcdatachannel: Chain to parent class constructed
And add a debug log statement.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2427>
2022-06-07 11:29:32 +00:00
Robert Mader
eb915b662a gstwaylandsink: Add support for the "render-rectangle" property
We already implement the `set_render_rectangle` videooverlay interface,
thus install the videooverlay property accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2479>
2022-06-06 14:36:39 +02:00
Robert Mader
8c3e33d494 gstwayland: Move reusable parts of the waylandsink into a library
In preparation for the new element `GstGtkWaylandSink`, move reusable
parts out of `GstWaylandSink` into the already exisiting but very
barebone library.

Notable changes include:
 - the `GstWaylandVideo` interface was dropped
 - support for `wl-shell` was dropped
 - lots of renaming in order to match established naming patterns
 - lots of code modernisations, reducing boilerplate
 - members were made private wherever possible

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2479>
2022-06-06 14:36:39 +02:00
Jan Alexander Steffens (heftig)
d86ad30be2 opencv: Allow building against 4.6.x
Replace the broken version checks with one modeled after
`GLIB_CHECK_VERSION`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2557>
2022-06-06 00:30:15 +02:00
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
Tim-Philipp Müller
962dc37d4f webrtc: fix build with older libnice versions
1) check for right macro name when checking for NICE_VERSION_CHECK

2) if libnice version is 0.1.18.1 this should not satisfy
   a NICE_VERSION_CHECK(0,1,19).

Fixes build with libnice 0.1.18.1 subproject checkout.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2499>
2022-05-26 18:17:49 +00:00
Philippe Normand
eefd793011 webrtc: Use new libnice API to get the candidate relay address
Corresponding libnice API added in:
https://gitlab.freedesktop.org/libnice/libnice/-/merge_requests/229 (0.1.19)
https://gitlab.freedesktop.org/libnice/libnice/-/merge_requests/232 (0.1.20)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Philippe Normand
08021caa73 webrtc: Ensure the NICE_CHECK_VERSION macro is available
This macro was introduced in libnice 0.1.19.1, so until we bump our libnice
dependency to 0.1.20 we have to vendor the macro.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Philippe Normand
c19319c777 webrtc: Refactor ICECandidateStats freeing logic to a dedicated function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Philippe Normand
dce8a7750d webrtcbin: Document IceCandidateStats and RTCIceCandidatePairStats
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
Philippe Normand
556ee45bfa datachannel: Notify low buffered amount according to spec
Quoting
https://www.w3.org/TR/webrtc/#dom-rtcdatachannel-bufferedamountlowthreshold

The bufferedAmountLowThreshold attribute sets the threshold at which the
bufferedAmount is considered to be low. When the bufferedAmount decreases from
above this threshold to **equal** or below it, the bufferedamountlow event fires.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2448>
2022-05-19 05:52:51 +00:00
Ludvig Rappe
26263c194e webrtc: Fix memory leak in icestream
Since both g_value_set_object() and g_weak_ref_get() takes a reference
there will be two new references to the GstWebRTCICE object when there
should be only one. g_value_take_object() has the same functionality as
g_value_set_object() but does not take a reference.

Without this change, the GstWebRTCICE object will be leaked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2333>
2022-04-29 21:52:43 +00:00
Thibault Saunier
4fd3886f5d qroverlay: Reset data_changed after we use the info
It was never reset so it was always TRUE once the data was changed!

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2299>
2022-04-27 15:09:47 +00:00
Thibault Saunier
1b31a2af45 qroverlay: Add a GstQROverlay meta
See documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2299>
2022-04-27 15:09:47 +00:00
Stéphane Cerveau
c77d07752a srtpdec: add counts in stats
In order to count the buffers which have been received and dropped for
decryption reason, add a stats to track it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2027>
2022-04-25 13:57:42 +00:00
Stéphane Cerveau
9d6a7dbdf3 rvsg: fix cairo include
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2276>
2022-04-23 00:00:23 +00:00
Sangchul Lee
c5b1eecb69 webrtcbin: Avoid access of freed memory
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2256>
2022-04-22 14:45:05 +00:00
Wonchul Lee
150db81287 dashsink: Unlock when failed to get content
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2242>
2022-04-20 09:07:29 +00:00
Xavier Claessens
b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00
Olivier Crête
2771490992 wpevideosrc: Give WebKit the keyboard, touch and pointer modifiers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2156>
2022-04-12 11:52:34 +00:00
Olivier Crête
41967e503c wpesrc: Convert from utf32 to support other keys
This makes all of the non-letter keys work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2156>
2022-04-12 11:52:34 +00:00
Olivier Crête
3ca61ae0d3 wpesrc: Initialize key event to 0
Otherwise, WebKit sees random modifiers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2156>
2022-04-12 11:52:34 +00:00
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