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>
H.266 NAL unit header syntax [1] is similar to H.265 NAL unit header syntax[2]:
```
H.265 H.266
+---------------+---------------+ +---------------+---------------+
|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7| |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| NALType | LayerId | TID | |F|U| LayerId | NALType | TID |
+-------------+-----------+-----+ +-------------+-----------------+
Where
* F: `forbidden_zero_bit`: f(1)
* U: `nuh_reserved_zero_bit`: u(1) only H.266
* LayerId: `nuh_layer_id`: u(6)
* NALType: `nal_unit_type`: u(6) in H.265 and u(5) in H.266
* TID: `nuh_temporal_id_plus1`: u(3)
```
NAL unit types have different values:
| NALType | H.265 | H.266 |
|----------|------------------------------------|---------------------------|
| VPS | HEVC_NAL_VPS(32) | VVC_VPS_NUT(14) |
| SPS | HEVC_NAL_SPS(33) | VVC_SPS_NUT(15) |
| PPS | HEVC_NAL_PPS(34) | VVC_PPS_NUT(16) |
| IRAP | BLA_W_LP(19)..HEVC_NAL_CRA_NUT(21) | IDR_W_RADL(7)..CRA_NUT(9) |
Implementation of `h266_video_type_find` is based on `h265_video_type_find` with
next differences:
- NAL unit header syntax for H.265 and H.266
- Diff NAL unit types values
- Avoid checking nuh_layer_id is zero. H.266 conformance test suite[3] contains examples with more than one layer.
This typefind was tested with H.266 conformance test suite [3]. Also, with the help of fluster[4],
with H.264 and H.265 conformance test suites to avoid regresions. Pending test vectors to fix:
- 8b422_H_Sony_4
- DEBLOCKING_E_Ericsson_3
[1] https://www.itu.int/rec/T-REC-H.266
[2] https://www.itu.int/rec/T-REC-H.265
[3] https://www.itu.int/wftp3/av-arch/jvet-site/bitstream_exchange/VVC/draft_conformance/draft6/
[4] https://github.com/fluendo/fluster/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7339>
In order to ensure all initial events (stream-start, caps, ..) are present on
pads that we expose, those various sticky events are propagated (from parsebin
to multiqueue output, from multiqueue output to exposed pads).
The problem was that the "hack" in `urisourcebin` to inform downstream elements
that the stream is parsed data and a collection will be present was only done in
one place : a probe on the output of parsebin ... but the stream-start could
potentially have already been propagated to the output pads before that.
In order to fix that, we make sure any pending sticky stream-start event is
updated before being propagated.
Fixes#3788
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7598>
Previously urisourcebin only allows stream-collections messages from adaptive
demuxers or sources to be posted.
This commit also allows the case where they come from a single parsebin. We
still want to prevent it in the case where they are multiple parsebins, since
that would require some form of aggregation to show a single/unified collection.
In order to avoid a regression with uridecodebin3 behavior, we also implement
support for GST_QUERY_SELECTABLE, so that uridecodebin3 can figure out whether
it should let GST_MESSAGE_STREAM_COLLECTION flow upwards (because app/user could
react on it) or whether it drops it in order for decodebin3 to do the collection
aggregation and posting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7597>
The presence (or not) of a collection on an input will determine whether events
will be throttled so that there are only forwarded when that input gets a valid
collection.
Therefore the input lock should be used.
In addition to that, we want to ensure that the application/user has a chance to
reliably (i.e. synchronously) specify what streams it is interested in by
sending a GST_EVENT_SELECT_STREAMS.
But we cannot allow anything to go forward until that message posting has come
back, otherwise we run in various races.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3872
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7594>
P010 uses 16 bits per pixel, with least significant being padding. This
code worked with Intel display driver since they roundup that value, but
does not work with the generic DRM helpers which also support NV15,
which does not have any padding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7580>
By setting the earliest time to timestamp + 2 * diff there would be a difference
of 1 * diff between the current clock time and the earliest time the element
would let through in the future. If e.g. a frame is arriving 30s late at the
sink, then not just all frames up to that point would be dropped but also 30s of
frames after the current clock time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7459>
There were two main issues:
The mix matrix was not protected with the object lock
The code was mistakenly assuming that after updating the mix matrix
a reconfigure event sent upstream would be enough to cause upstream to
send caps again, and the converter was only reconstructed in ->set_caps.
That was not actually enough, as if the new matrix didn't affect the
number of input / output channels there was no reason for upstream to do
anything after getting the unchanged caps.
The fix for this was to have ->transform also recreate the converter
when needed, with the added subtlety that depending on the mix matrix
the element could be set to passthrough. This means that when setting
the mix matrix the converter also had to be recreated immediately to
check if the element had to be switched back to non-passthrough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7363>
I didn't find the behavior and purpose of streamsynchronizer documented
or intuitive. Eventually I got Edward to explain it to me, which was
very helpful. Now I'm contributing some docs so that the next person
doesn't have to figure it out by asking around and hoping for an answer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7084>
When glupload generates sink caps based on src caps after determining upload method, src
caps may only contain RGBA format.
In this case, the raw caps on the sink pad generated by glupload will only contain the
RGBA format, which will cause caps negotiation fail, because the filter caps used for
negotiation by the upstream element may only contain other formats, such as xBGR, etc.
Add the formats supported by #GstGLMemory to raw caps to ensure that caps negotiation
succeeds.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7061>
release_frame() can be useful for manually dropping frames without posting QoS messages like finish_frame() would.
Matches the same kind of API on the decoder side of things.
Modifies the behaviour of release_frame() to make sure events from released frames are stored as 'pending'
and pushed before the next non-dropped frame. This is needed because now release_frame() can be called outside of
finish_frame(), so we would potentially just lose events and bad things would happen.
drop_frame() was also added to match the decoder API. It functions almost identically to finish_frame() without a buffer
attached to the frame, except instead of immediately pushing the frame's events, it will store them as pending.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7190>
In case when conn->input_stream is NULL and glib was built with
"glib_checks" enabled, g_pollable_input_stream_read_nonblocking()
returns -1, but does not set the "err".
The call stack:
read_bytes() ->
fill_bytes() ->
fill_raw_bytes()
The return value -1 passed up to read_bytes() and incorrectly
processed there after "error:" label.
This changes the return value to EINVAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7210>
glCheckFrameStatus() can fail by returning 0, and otherwise return a
status. Fix the trace to make it clear when we get an unkown status
compare to having an error, in which case we also trace the error code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7291>
videoscale does not have convert function, so remove the convert
description in it's classification. Otherwise, if we want use
autovideoconvert to convert colorsapce, autovideoconvert will select
videoscale to do convert and this will cause to fail.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7215>
