Change locking around group deactivation to avoid deadlocks
when shutting down exactly as a buffering message arrives.
The PLAYBIN3_LOCK now protects the active field of the
source group. Everything else is still protected by the
source-group-lock.
Also properly protect group switching operations with
the PLAYBIN3_LOCK everywhere.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1049>
Previously suspended medias immediately reached the UNPREPARED state
without going through the media's unprepare() vfunc. This didn't allow
the media subclass to do any additional cleanup, and for example the
shutdown-eos property of GstRTSPMedia was ignored.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
Windows doesn't support fork so every test will be performed in
one process. So the test_meta_custom_transform() is being
failed because "test-custom" custom meta is being used/defined in
another test test_meta_custom() as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1086>
The GST_VIDEO_DECODER_ERROR() should be used only for robust/error-resilient
decoding purpose. Any other error codes such as not-negotiated or flushing
should be returned without modified for upstream to be able to handle
it immediately. (for example, application might want to try other
decoder element on not-negotiated)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1070>
tests/check/meson.build uses the openjpeg_dep variable
unconditionally, and the subdir_done() is useless anyway, since the
plugin is only built if openjpeg_dep.found() is true. Fixes:
..\tests\check\meson.build:23:0: ERROR: Unknown variable "openjpeg_dep".
In particular, this fixes the build on UWP since we disable openjpeg
explicitly in Cerbero when building for UWP.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1069>
handle_*_request() functions were all retrieving the session media from
the session by calling gst_rtsp_session_get_media () which is a transfer-none
call. If a session timeout happens at that time, the session media may get freed
making the pointer invalid..
Fixes#757
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
Instead of assuming that the PTS of a keyframe is the lowest PTS of a
GOP, wait until the DTS has passed this PTS and take the minimum PTS up
to that point. That way the minimum PTS of a GOP can be determined, at
least for closed GOP streams. Open GOP streams still can't be handled
properly.
By knowing the minimum PTS of each GOP, keyframes can be requested at
the correct time relative to the GOP (and thus fragment) start and
fragment overflow calculations can calculate the correct durations of
the GOPs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
In an embedded system where all services run as seperate users it is
useful to have the gstreamer registry readable by all so it can be
re-used, in similar manner as a host system where one user have seperate
applications running but all share same registry.
To make this possible introducing GST_REGISTRY_MODE for adjusting the
changing mode of the registry binary when finishing up with the
temporary file (which has restricted access).
Fixes: #692
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/825>
If the query has already been destroyed at this point, GST_IS_QUERY will
read garbage, can return false and we will try to unref it again.
Instead, make note of whether the item is a query when we dequeue it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1029>
boolean return value is not sufficient for representing the reason
of error in most cases. For instance, any errors around new_sequence()
would mean negotiation error, not just *ERROR*.
And some subclasses will allocate buffer/memory/surface on new_picture()
but it could be failed because of expected error, likely flushing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1019>
boolean return value is not sufficient for representing the reason
of error in most cases. For instance, any errors around new_sequence()
would mean negotiation error, not just *ERROR*.
And some subclasses will allocate buffer/memory/surface on new_picture()
but it could be failed because of expected error, likely flushing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1019>
boolean return value is not sufficient for representing the reason
of error in most cases. For instance, any errors around new_sequence()
would mean negotiation error, not just *ERROR*.
And some subclasses will allocate buffer/memory/surface on new_picture()
but it could be failed because of expected error, likely flushing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1019>
warning C4003: not enough arguments for function-like macro invocation 'warning'
G_STMT_END macro is extended to the below form with MSVC
__pragma(warning(push)) \
__pragma(warning(disable:4127)) \
while(0) \
__pragma(warning(pop))
So MSVC preprocessor will extend it further to
__pragma(VBI_CAT_LEVEL_LOG(push)) ...
Should rename warning() debug macro function therefore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1018>
libgudev is a problematic dependency, particularly in sandboxed
environments, such as flatpak.
This patch implements a way to get the available VA devices using
brute-forced traverse of /dev/drm/renderD* directory. Thus usable in
those sandboxed environments.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1027>
When move the libgstva, libgudev dependency was moved as part of the
library, though it's not use by the library but the plugin. This patch
moves back libgudev dependency to the plugin.
Also HAVE_LIBDRM is move to the library which is the one who use it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1027>
Some decoding APIs support delayed output for performance reasons.
One example would be to request decoding for multiple frames and
then query for the oldest frame in the output queue.
This also increases throughput for transcoding and improves seek
performance when supported by the underlying backend.
Introduce support in the vp9 base class, so that backends that
support render delays can actually implement it.
Co-authored by Seungha Yang <seungha@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/987>
Since 0d95d9258b we respect the asset stream-id in `GESUriSource` so
we can not work with unknown or broken stream ID in the assets.
We just ignore them, warning about it and we should fix that in
demuxer so they don't expose pad without providing a stream id for them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1001>
This patch contains two updates:
1. Instead of checking for dependency already checked just to verify a
version, we use the dependency version API.
2. Update the deprecated function get_pkgconfig_variable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/997>
It's possible to have installed MediaSDK environment
package (libmfx-dev in Debian) without libva environment package. This
setup will lead to a breakage of meson configuration.
The fix is to get the libva's driver directory variable after the
dependency is validated as found.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/998>
When using the following setup (the error can be reproduced using
simpler sender pipelines), the receiver resynchronises the clock on RTCP
packets. The effect was that a couple seconds were cut out of the
playback because an initial RTCP packet was dropped.
When sending out all RTCP packets (setting sync=FALSE on the RTCP
updsink), the playback is fine.
This syncs rtpsink with rtpsrc (where this property was already set).
gst-launch-1.0 filesrc location=899-en.mp3 \
! mpegaudioparse \
! mpg123audiodec \
! audioconvert \
! audioresample \
! avenc_g722 \
! rtpg722pay
! rtpsink uri=rtp://239.1.2.3:1234
gst-launch-1.0 uridecodebin rtp://239.1.2.3:1234?encoding-name=G722 \
! autoaudiosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/993>
Since the base class now does the parsing, there is no need
to reproduce that code in all the subclasses, just pass the attributes
which are the only relevant bit anyway.
Also, only store the direction if the subclass accepted the caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
If we are not receiving a sync-point for a very long time, we need to
keep asking for them. The request-sync-point logic keeps track of how
many keyunitrequests we are allowed to send, but that would not matter
if we don't keep asking.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/930>
When there are elements between the demuxer and the muxer that
introduce an offset to the running time, or when offsets are
set on pads by the application, this shift must be taken into
account when calculating the final pts_adjustement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
mpegtsmux can receive SCTE sections from two origins: events
created by the application, and events forwarded downstream by
mpegtsdemux, containing sections that may not have been fully
parsed, and additional data to help tsmux translate times to
the correct domain, both for requesting keyframes and calculating
an accurate pts_adjustment.
The complete approach is documented further in a comment above
the relevant function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
Instead of modifying the splice times in the incoming sections
to running time and expecting eg mpegtsmux to convert those back
to its local PES time domain, which might be impossible when
those splice times are encrypted or the specification is extended,
transmit the needed information to the muxer as separate fields in
the event:
* A pts offset field can be used by the muxer in order to calculate
a final pts_adjustment
* A rtime_map can be used by the muxer to determine the correct
running times at which it should request keyframes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
Makes it possible to support passing SCTE 35 cue points from
demuxer to muxer, while preserving correct timing.
This will also improve ex nihilo cue points injection, as splice
times and durations are now interpreted as running time values,
and may trigger key unit requests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
Main differences with previous setup are:
- No manifest creation
- gst-indent is executed only when the bot is assigned (instead of the manifest task)
- Cerbero jobs are triggered in the cerbero repo
- Remove cerbero and android related files as they now are in cerbero
itself.
- Update `container.ps1` to the new file layout
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/891>
glib-networking fixed some issues when building with older libssl version
in the 2.68 release, update the wrap file to use the newer version.
In particular this fixes building on Ubuntu 16.04 with:
meson --wrap-mode=forcefallback
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/247>
