People might have GST_TRACERS=leaks set in their environment
by default, which will now trigger criticals during the build
when calling g-ir-scanner, because we unset GST_PLUGIN_SYSTEM_PATH
so that the scanner doesn't load any plugins.
Fixes#4093
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8121>
There is no reason why we should mandate people to "at least" use the static
sink pad. This caused issues, like mandating that it should always have valid
content linked to it (problematic in case of upstream stream changes).
Instead we only use it if it's actually linked to, in which case it gets added
to the list of inputs.
This actually simplifies the code too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7949>
When checking whether a no-longer used output could be re-used for another slot,
we only want to do that for streams which are not still used.
Otherwise we end up potentially re-assigning a demuxer stream to a completely
different one
Furthermore, if we *are* re-using an output slot, indicate what the replacement
GstStream will be so slot matching can work properly (which can happen in the
case of demuxers which add/remove all pads even if only a single one changed)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7949>
We get loads of warnings when parsing videos from users:
gsth264parser.c:1115:gst_h264_parser_parse_user_data_unregistered: No more remaining payload data to store
gsth264parse.c:646:gst_h264_parse_process_sei:<h264parse0> failed to parse one or more SEI message
Those are raised because of unregistered SEI without user data.
The spec does not explicitly state that unregistered SEI needs to have
data and I suppose the UUID by itself can carry valuable information.
FFmpeg also parses and exposes such SEI so there is no reason for us no
too as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7931>
Convert segment to TIME format immediately instead of waiting for
_chain() to be called. This fixes converted segment never being pushed
downstream.
Fix the convert function that was copying some fields in the wrong
direction. Add fast copy if segment is already in TIME format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7977>
At least some Mesa drivers (see llvmpipe_query_dmabuf_modifiers()) don't
initialize the provided external_only array, so some format/modifier
combinations could get incorrectly marked as only external.
Make sure we zerofill the array before passing it to
eglQueryDmaBufModifiersEXT().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>
* if downstream provides us with a pool that can allocate DMA buffers,
gldownload wraps it into a GL buffer pool that uses
glEGLImageTargetTexture2DOES() to import dmabuf into textures
* upstream GL elements can allocate from that pool
* gldownload unwraps DMA buffers from incoming GL buffers and passes
them downstream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>
We only want to switch to a selection of an output collection if all streams are
present.
This was previously only done in one place (when triggering by new incoming
streams) but not when triggered by user/application.
Avoid this by moving the check to handle_stream_switch()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7941>
A GST_VIDEO_FORMAT_* could be mapped to only one pair of DRM_FORMAT_*
and DRM_FORMAT_MOD_*. Until now only DRM_FORMAT_MOD_LINEAR was used.
To be able to add other modifiers add a modidier field in struct FormatMap.
Create a helper functions the allow turning a GstVideoFormat into a pair
of DRM fourcc and modifier and vis-versa.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7849>
Variable f1 is never used, so just skip that loop for now.
The test has never actually tested actual resampling because of
that bug it seems, and the test fails if fixed to actually resample.
For now we just avoid the pointless 126*12 pipelines that were just
testing the same thing (nothing) over and over again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7827>
Combine the appsrc and appsink settings into one place and ensure that
the appsrc will output a TIME segment, to avoid incorrect segment format
criticals in some situations.
The D3D11 path was already setting the segment format correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7839>
The gst_dep.get_variable('libexecdir') may fail in some scenarios
(e.g. building a module alone inside an uninstalled devenv) and
it shouldn't really be reached in the first place if docs are
disabled via options.
Also to avoid confusing meson messages when cross-compiling or
doing a static build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7818>
Don't assume that video/x-raw caps means buffers are mappable
or can be processed by videoconvert and friends. Only plug
those converters for real system memory, and treat other
memory capsfeatures as hardware surfaces
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7741>
There is the possibility than an element/code/helper creates an identical
`GstStream` (same type and stream-id) instance instead of re-using a previous
one.
For those cases, when detecting whether a `GstStream` is already present in a
collection, we need to do more checks than just comparing the pointer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7716>
If we can't get the current caps when receiving a stream-start, that's fine,
they can/will be provided by other means at a later time.
What we definitely should not do is provide the starting caps of the chain,
which are potentially completely different from the end ones (like for example
`application/x-rtp`)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7716>
If an encoder supports multiple codecs (a bin wrapping/auto-plugging encoders)
then its src pad template caps might list the supported codecs. Without this
patch the selected parser would be the one corresponding to the first codec,
leading to caps negotiation error later on. The proposed fix is to check the
media type on the parser candidates sink pad templates according to the
requested encoded format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7670>
Some servers (e.g. Axis cameras) expect the client to propose the encryption
key(s) to be used for SRTP / SRTCP. This is required to allow re-keying so
as to evade cryptanalysis. Note that the behaviour is not specified by the
RFCs. By setting the 'client-managed-mikey-mode' property to 'true', rtspsrc
acts as follows:
* For a secured profile (RTP/SAVP or RTP/SAVPF), any media in the SDP
returned by the server for which a MIKEY key management applies is
elligible for client managed mode. The MIKEY from the server is then
ignored.
* rtspsrc sends a SETUP with a MIKEY payload proposed by the user. The
payload is formed by calling the 'request-rtp-key' signal for each
elligible stream. During initialisation, 'request-rtcp-key' is also
called as usual. The keys returned by both signals should be the same
for a single stream, but the mechanism allows a different approach.
* The user can start re-keying of a stream by calling SET_PARAMETER.
The convenience signal 'set-mikey-parameter' can be used to build a
'KeyMgmt' parameter with a MIKEY payload.
* After the server accepts the new parameter, the user can call
'remove-key' and prepare for the new key(s) to be served by signals
'request-rtp-key' & 'request-rtcp-key'.
* The signals 'soft-limit' & 'hard-limit' are called when a key
reaches the limits of its utilisation.
This commit adds support for:
* client-managed MIKEY mode to srtpsrc.
* Master Key Index (MKI) parsing and encoding to GstMIKEYMessage.
* re-keying using the signals 'set-mikey-parameter' & 'remove-key' and
then by serving the new key via 'request-rtp-key' & 'request-rtcp-key'.
* 'soft-limit' & 'hard-limit' signals, similar to those provided by srtpdec.
See also:
* https://www.rfc-editor.org/rfc/rfc3830
* https://www.rfc-editor.org/rfc/rfc4567
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7587>
This allows the stream to drive the buffers submitted to the display server.
If the application does not receive frame events for a period of time due to
minimization or tty switch for example, instead of waiting to process and
then catching up when frame events resume, the stream will resume instantly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7691>
There is no requirement for a base DRM format to be supported by libgstvideo
in order to be uploaded to. Don't limite to DRM fourcc that have a libgstvideo
format mapping. This notably enabled AFBC support, which uses an opaque based
format that does not have a linear definition. This also adds R8/RG88 and
simimlar other formats that are not yet mapped in libgstvideo.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7689>
We want to ensure the stream-collection is present on the pad (as a sticky
event) before we expose the pad.
This is more reliable since it will ensure it is present before any other event
is pushed through.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>
