We should not reset the input/output_frame_count when some configure
changes. For example, the if resolution changes, the current way just
resets the frame count and make the PTS of the output buffer restart
from the original PTS of the first frame. That causes a lot of QOS
event and drop all the new frames.
We should only reset them when encoder start().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2489>
When collection is updated, decodebin3 exposes pad first and then
streams-selected message is posted.
The condition can cause a situation where playbin3 links non-existing
combiner/playsink pads (since streams-selected is not posted yet) with
new decodebin output pad. This commit will re-check selected/active
streams condition on pad-added and reconfigure output if needed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2482>
In case of per features registration such as the
customizable gstreamer-full library, each
element should check that the soup library can be loaded to
facilitate the element registration.
Initialize the debug category properly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2349>
d3d11screencapture can miss a cursor shape to draw or draw an outdated cursor shape.
- AcquireNextFrame only provides cursor shape when there is one update
- current d3d11screencapture skips cursor shape when mouse is not drawn
So, if a gstreamer application uses d3d11screencapture with cursor initially not drawn
"show-cursor"=false and then switches this property to true, the cursor will not be
actually drawn until AcquireNextFrame provides a new cursor shape.
This commit makes d3d11screencapture always update the cursor shape information, even
if the mouse is not drawn. d3d11screencapture will always have the latest cursor shape
when requested to draw it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2485>
zlib is required, and if it isn't found it is checked several ways and
then forced via subproject(). This code was added in commit
b93e37592a, to account for systems where
zlib doesn't have pkg-config files installed.
But Meson already does dependency fallback, and also, since 0.54.0, does
the in-between checks for find_library('z') and has_header('zlib.h') via
the "system" type dependency. Simplify dependency lookup by marking it
as required, which also makes sure that the console log doesn't
confusingly list "not found".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2484>
The pool process function may poll and get the resolution-change event
whenever it is not possible to share our buffers. This typically happen
when downstream does not support GstVideoMeta.
Not handling this would cause the decoder thread to exit silently and the
pipeline to stall.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2457>
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>
* Enhance debug log to print human readable D3D11_FORMAT_SUPPORT flags
value, instead of packed numeric flagset value.
* Only device supported format will be added to format table.
Depending on device feature level (i.e., D3D9 feature devices),
16bits formats will not be supported. Although there might be formats
we deinfed but not supported, it will not be a major issue in practice
since our D3D11 implementation does not support legacy devices already
(known limitation) and also old d3dvideosink will be promoted in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2441>
Output may attemp to set the width and height to zero values if
caps has no such information, which will cause capture get invalid
dimensions. Then decoder reports negotiation failure.
So need to set default resolution if caps has no such information.
Real values can be set again until source change event is signaled.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2400>
Now it uses the JPEG parser in libgstcodecparsers, while the whole
code is simplified by relying more in baseparser class for tag
handling.
The element now signals chroma-format and default framerate is 0/1,
which is for still-images.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1473>
While current and future LoongArch machines that are supposed to run
GStreamer all support unaligned accesses, there might be future
lower-end cores (e.g. the embedded product line) without such support,
and we may not want to penalize these use cases.
So, mark LoongArch as not supporting unaligned accesses for now, and
hope the compilers do a good job optimizing them. We can always flip
switch later.
Suggested-by: CHEN Tao <redeast_cn@outlook.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2443>
It is valid to have the padding set to 1 on the first packet and it
happens very often from TWCC packets coming from libwebrtc. This means
that we were totally ignoring many TWCC packets.
Fix test that checked that a first packet with padding was not valid and
instead test a single twcc packet with padding to check precisely what
this patch was about.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2422>
When processing the first event after probing the
file and being activated, requeue sticky events
as there's no requirement that demuxers send tag
and other events again after a seek - that's
why they're sticky.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2432>
- Consistently unref the chained buffer at the end of the chain
function, if we're not handing it off to `gst_pad_push`. This avoids a
few buffer leaks in the error paths in `_chain` and `_push_history`.
- When mapping the video frame fails, return a flow error instead of
crashing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2428>
If we just break the loop, we might run into the `gop != NULL` assert
that follows it. Rather, exit immediately with flushing flow.
Also use this flushing mechanism when we release a pad. This avoids
having an extra flag.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1030>
Background:
Whenever a caps event is received by appsink, the caps are stored in the
same internal queue as buffers. Only when enough buffers have been
popped from the queue to reach the caps, `priv->sample` gets its caps
updated to match, so that they are correct for the following buffers.
Note that as far as upstream elements are concerned, the caps of appsink
are updated immediately when the CAPS event is sent. Samples pulled from
appsink retain the old caps until a later buffer -- one that was sent by
upstream elements after the new caps -- is pulled.
The race condition:
When a flush is received, appsink clears the entire internal queue. The
caps of `priv->sample` are not updated as part of this process, and
instead remain as those of the sample that was last pulled by the user.
This leaves open a race condition where:
1. Upstream sends a new caps event, and possibly some buffers for the
new caps.
2. Upstream sends a flush (possibly from a different thread).
3. Upstream sends a new buffer for the new caps. Since as far as
upstream is concerned, appsink caps are the new caps already, no new
CAPS event is sent.
4. The appsink user pulls a sample, having not pulled before enough
samples to reach the buffers sent in step 1.
Bug: the pulled sample has the old caps instead of the new caps.
Fixing the race condition:
To avoid this problem, when a buffer is received after a flush,
`priv->sample`'s caps should be updated with the current caps before the
buffer is added to the internal queue.
Interestingly, before this patch, appsink already had code for this, in
gst_app_sink_render_common():
/* queue holding caps event might have been FLUSHed,
* but caps state still present in pad caps */
if (G_UNLIKELY (!priv->last_caps &&
gst_pad_has_current_caps (GST_BASE_SINK_PAD (psink)))) {
priv->last_caps = gst_pad_get_current_caps (GST_BASE_SINK_PAD (psink));
gst_sample_set_caps (priv->sample, priv->last_caps);
GST_DEBUG_OBJECT (appsink, "activating pad caps %" GST_PTR_FORMAT,
priv->last_caps);
}
This code assumes `priv->last_caps` is reset when a flush is received,
which makes sense, but unfortunately, there was no code in the flush
code path resetting it.
This patch adds such code, therefore fixing the race condition. A unit
test demonstrating the bug and testing its behavior with the fix has
also been added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2413>
This transition is meant to be very similar to crossfade, but
instead of fading out the background video at the same time as the
foreground fades in, the background video stays at 100% opacity
during the whole transition.
This essentially "restores" the old crossfade behaviour that was changed in:
eb48faf342
but using a new type enum, so that both behaviours are available,
letting applications choose.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2385>
The timestamp in the tfdt refers to the first trun box and if there are
multiple trun boxes then the distance between the first timestamps will
grow.
At some point this distance reaches a threshold and triggers the
resetting of the first sample's timestamp of this trun box to be reset
to the tfdt.
This threshold is implemented for files where there is a jump in the
timeline between fragments and where this can be detected via a jump
between the end timestamp of the previous fragment and the tfdt of the
next. This behaviour is preserved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2409>
NVIDIA GPUs have undocumented limitation regarding minimum resolution
and it can be queried via a NVDEC API. However, since we don't want to
bring CUDA/NVDEC API into D3D11, use hardcoded values for now
until we find a nice way for capability check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2406>
Various elements are assuming that the pointer matches a pad template
they know about, and also randomly created pad templates might be
missing some important information that is necessary to create a valid
pad.
For example, creating a new pad template for audiomixer's sinkpad
without providing the correct GType would cause audiomixer to create a
GstAggregatorPad. That will then later fail spectacularly because it
assumes that it got a GstAudioAggregatorPad.
Passing a pad template that does not belong to the element class in here
will easily lead to undefined behaviour.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2410>
If for some reason the encoder produces frames with a pts higher than
the input one, we were dropping all the video encoder frames and ended
up crashing when trying to access the pts of a NULL pointer returned by
gst_video_encoder_get_oldest_frame().
I hit this scenario by feeding a decreasing timestamp to vp8enc which
seem to confuse the encoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2405>
Until March 2022, the FFmpeg MXF muxer would write the various index table
segments with the same instance ID, which should only be used if it is a
duplicate/repeated table.
In order to cope with those, we first compare the other index table segment
properties (body/index SID, start position) before comparing the instance
ID. This will ensure that we don't consider them as duplicate, but can still
detect "real" duplicates (which would have the same other properties).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2407>
If the stream chroma doesn't match with any video format in the source
caps template (generated from va config surface formats) instead of
return unknown, return the first available format in the template,
assuming that the driver would be capable to do color conversions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2404>
Use newly added gst_h265_parser_identify_and_split_nalu_hevc()
method to handle broken streams where packetized NAL unit
contain start code prefix in it.
It's obviously wrong stream but we know how to work around it
and even need to support such broken streams since
stateless decoder implementations are being a primary
decoder element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2394>
Add gst_h265_parser_identify_and_split_nalu_hevc() method to
handle a case where packetized stream contains start-code prefix.
This new method behaves similar to exisiting gst_h265_parser_identify_nalu_hevc()
but it will scan start-code prefix to split given data into
NAL units.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2394>
Instead of using a hard-coded list of preferred formats according the
chroma type, now if now caps are pre-negotiated, from template caps
will choose the first format with the same chroma type. If
pre-negotiated, then it will choose the first format, with same chroma
type, from the first caps structure.
Also all the decoders will check if GST_VIDEO_FORMAT_UNKNOWN is
returned, failing the negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2351>
Hantro H1 and Rockchip VEPU2 drivers will pad the width/height to a
multiple of 16. In order to obtain the right JPEG size, the image needs
to be cropped using the S_SELECTION API. This support is added as best
effort since older drivers may emulate this by looking at the capture
queue width/height.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2329>
gst_value_serialize() does more than what's needed to printf-ing
especially when given GValue is already string. Just print string
value as-is without gst_value_serialize() to avoid unreadable
string print, especially for multi-bytes character encoding cases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2387>
V4L spec now requires decode_params flags to be set in accordance to the
frame's type. In particular this is required by H.264 decoder of NVIDIA
Tegra SoC to operate properly. Set the flags based on type of parsed
slices.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1757>
* Remove fields no longer used, or that can be replaced by smaller code
* Rename "channels" to a more meaningful "input pads"
* Directly handle/use combiner pads in the combiners instead of on the playbin3
main structure
Remove the corresponding combiner sinkpad whenever a uridecodebin3 source pad
goes away
* If used, store the corresponding combiner sink pad in the SourcePad helper
structure
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2384>
GstD3D11ScreenCapture object is pipeline-independent global object
and the object can be shared by multiple src elements,
in order to overcome a limitation of DXGI Desktop Duplication API.
Note that the API allows only single capture session in a process for
a monitor.
Therefore GstD3D11ScreenCapture object must be able to handle a case
where a src element holds different GstD3D11Device object. Which can
happen when GstD3D11Device context is not shared by pipelines.
What's changed:
* Allocates capture texture with D3D11_RESOURCE_MISC_SHARED for the
texture to be able to copied into other device's texture
* Holds additional shader objects per src element and use it when drawing
mouse
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1197
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2366>
mp4mux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.
By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
mxfmux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.
By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
mpegtsmux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.
By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
flvmux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.
By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
Otherwise setting the srcpad caps based on the sinkpad caps event will
already push a segment event downstream before the upstream segment is
known.
If the upstream segments are just forwarded when the upstream segment
event arrives this would result in two segment events being sent
downstream, of which the first one will usually be simply wrong.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
Baseclass calls get_preferred_output_delay() in a chain of
sequence header parsing and then new_sequence() is called
with required DPB size (includes render-delay) information.
Thus latency query should happen before the sequence header
parsing for subclass to report required render-delay accordingly
via get_preferred_output_delay() method.
(e.g., zero delay in case of live pipeline)
This commit is to fix wrong liveness signalling in case of
upstream packetized format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2363>
Baseclass calls get_preferred_output_delay() in a chain of
sequence header parsing and then new_sequence() is called
with required DPB size (includes render-delay) information.
Thus latency query should happen before the sequence header
parsing for subclass to report required render-delay accordingly
via get_preferred_output_delay() method.
(e.g., zero delay in case of live pipeline)
This commit is to fix wrong liveness signalling in case of
upstream packetized format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2364>
In the case where not all streams have received any data, growing the interleave
by only 100ms is too restrictive and would cause some (valid) mpeg-ts streams to
hang.
Bump up the interleave growth rate for those use-cases to 500ms per input (still
up to the limit of 5s).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2370>
If there weren't any moved/dirty regions in the captured frame, the
viewport of the ID3D11DeviceContext would be left at whatever previous
value it had, which could lead to the cursor being drawn in a wrong
position and/or in an incorrect size.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2362>
Make all codecs consistent so that subclass can know additional DPB
size requirement depending on render-delay configuration regardless
of codec. Note that render-delay feature is not implemented for AV1
yet but it's planned.
Also, consider new_sequence() is mandatory requirement, not optional
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2343>
As implemented, we only support OpenGL 3 API from version 3.2. Though, there
is no issue enabling GLSL 1.30 even if we are going to restrict our API usage
to 2. This allows using texelFetch() on OpenGL 3.0 and 3.1 drivers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2190>
Since the addition of tiling format with subsampled tile size
(NV12_16L32S), getting the tile width/height shifts and tile
size have become more complex. Add a helper to extract and
scale this information for the selected plane and format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2190>
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>
Mixing C loops with switch statements is a bad idea as break has a
different meaning in both. Breaking inside the switch statements wrongly
caused further loop iterations.
Instead use goto to get out of the loop and continue to do another loop
iteration, and never ever use break except for the end of a case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2336>
Some streams have 2 PMT sections in a single TS packet. The first one is "valid"
but doesn't contain/define any streams. That causes an unrecoverable issue when
we try to activate the 2nd (valid) PMT.
Instead of doing that, pre-emptively refuse to process PMT without any streams
present within. We still do post that section on the bus to inform applications.
Fixes#1181
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2310>
In push mode (streaming), if the received chunk buffer size from _chain is bigger
than output buffer size, the flags of the divided-buffers are propagated to the
DISCONT flag from first received chunk buffer. This unexpected buffers contained DISCONT
flags are abnormally transformed when changing the sampling rate by audioresample element.
So unset unnecessary DISCONT flag before pad_push().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2305>
Previously, we only added it when actually performing synchronization
based on the NTP time.
The information can be useful downstream in other situations too, and
we can compute a NTP time as soon as we get a sender report with the
relevant information.
Co-authored-by: Mathieu Duponchelle <mathieu@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2252>
The decision to store the input buffer depends on whether extensions
are to be added to the output buffer, I assume as an optimization.
This creates an issue for subclasses that call negotiate(), where
header_exts is actually populated, from their handle_buffer()
implementation: at chain time, no header extension has been negotiated
yet, which means that we don't add extensions to the first batch of
buffers that comes out.
Keep track of whether negotiate has been called (this is different
from the negotiated field) and always store the input buffer until
then. This fixes the issue while largely preserving the optimization.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2304>
The previous iteration of the code was inferring the type of the
frame by looking at the overall size of the gst-payloaded packet.
It is more robust to actually parse the payload and look at the
actual data buffers it contains.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
regardless of whether they are input as individual buffers or
buffer lists.
The ONVIF specification requires all packets to hold the extension,
it makes no sense to behave differently when handling buffer lists.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2303>
Pipeline such as:
gst-launch-1.0 -vf videotestsrc ! video/x-raw,format=NV12,colorimetry=\(string\)bt709 \
! videoscale ! video/x-raw,format=I420 ! fakesink
Always trigger a error:
ERROR video-info video-info.c:556:gst_video_info_from_caps: no width property given
Because it is called before the fixate_size(), the src caps' resolution
may be absent or not fixed. That causes that the src video info can not
be created correctly and we can not inherit the colorimetry and chroma-site
from the input caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2289>
Fixing this pipeline:
gst-launch-1.0 filesrc location=sample.png ! pngdec ! videorate ! fakesink
- videorate receives a single buffer with pts = 0, duration = invalid;
- then it receives eos triggering this buffer to be pushed downstream;
- the pushing code was assuming that a duration was set, which is
impossible as we received a single buffer and no output framerate was
set either. So the best we can do is to push the buffer without
duration.
Fix#1177
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2296>
The va pool is used for GPU side surface/image, its alignment should
not be changed arbitrarily by others. So we decide not to expose the
GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT flag anymore.
Instead, user can call gst_buffer_pool_config_set_va_alignment() to
set its surface/image alignment.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2282>
According to spec:
color range equal to 0 shall be referred to as the studio swing
representation and color range equal to 1 shall be referred to as
the full swing representation.
The current status is just the opposite.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2288>
GAP events flagged with MISSING_DATA are transformed into GAP buffers
flagged with CORRUPTED.
In these cases, it is preferable to simply keep rendering the previous
buffer (if there was one) instead of flashing the pad in and out of
view.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/708>
When the GAP event was flagged with MISSING_DATA, subclasses
may want to adopt a different behaviour, for example by repeating
the last buffer.
As we turn these gap events into gap buffers, we need to flag
those, we do so with a new custom meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/708>
Returning TRUE from the `transform_meta` function tells
GstBaseTransform to copy the meta into the new buffer. If videoscale
has already transformed a meta by scaling it, it should always return
FALSE to avoid duplicating the meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1630>
Meson generates a gdbinit file that will automatically load gstreamer
script. However that script uses a helper python module that needs
PYTHONPATH to be pointing into the right location in the source
tree to be able to find gst_gdb.py.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1796>
When the sink is configured to create sockets with an explicit bind
address, then the created socket gets set to the udp_socket field
irregardless of whether the bind address indicated that the socket
family should be IPv4 or IPv6. When binding to an IPv6 address, this
results in the following error:
gstmultiudpsink.c:1285:gst_multiudpsink_configure_client:<rtcpsink>
error: Invalid address family (got 10)
This patch adds a check of the address family being bound to and sets
the created socket to used_socket or used_socket_v6, accordingly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1551>
RFC 8216 6.3.3 "Playing the Media Playlist File" : states that for live media
playlists "the client SHOULD NOT choose a segment that starts less than three
target durations from the end of the Playlist file"
This is an off-by-one error. Since we are looking for the "index" of the
segment, we need to subtract 1 from the searched position.
Ex: For a playlist with 12 entries, we want to start playback on the 9th segment
... which is at index 8.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2259>
When we fixup src caps, the current way of handling the HDR fields is not
correct.
1. We trim the HDR fields only when the input caps is not a subset of the
fixup src caps. But in fact, the input caps with HDR fields such as the
"mastering-display-info" can possibly be the subset of the fixup src caps,
if they have all same other fields.
2. We always copy the colorimetry from input caps to src caps if it is
absent. But when hdr-tone-mapping is enabled, the HDR->SDR conversion makes
the colorimetry change. We should use downstream's setting, or just use the
default colorimetry of SDR.
We changes to:
1. If hdr-tone-mapping is enabled, we trim all HDR fields and add a correct
colorimetry.
2. Copy colorimetry from input if it is still absent.
3. Consider the subset replacement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2244>
Since d0133a2d11 "videoconvert: Allow
passthrough for ANY caps features" videoconvert will always claim that
it supports any kind of memory which is true in very specific case (when
it is running in passthrough mode). To get elements that autoplug
converters depending on the caps running in the pipeline (like
autovideoconvert), we need to have converters no lie about what they can
do when queried `accept_caps` or `query_caps`.
This still accepts any caps feature as before but it introduces
a restriction in the way we handle memory capsfeatures.
We keep previous behaviour in videoconvert and videoscale.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>
Now that videoconvert and videoscale's are both based on
GstVideoConverter and are using the exact same code, it makes much more
sense to have one element doing the two operation, and it can be
more efficient in some cases (one single path for both operations).
This removes the `videoscale` and `videoconvert` plugins but keeps the element
but makes them also do both operations (adding some APIs to each element).
There is a small change in API for the `videoscale:dither` property which
was previously a totally unused boolean, it is now an enum and is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>
The RTCP SR packet might be without SDES in case of a reduced-size RTCP
packet. For syncing purposes the CNAME is needed but it might be known
already from an earlier RTCP packet or out of band, via the SDP for
example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
The format of the caps fields is
ssrc-(SSRC_VALUE)-(ATTRIBUTE_NAME)=(ATTRIBUTE_VALUE)
.
Parsing of the attributes from the caps into the SDP is not implemented
as this depends not only a single stream's caps but on the whole rtpbin
configuration.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
Found via an analyzed build for Clang. Specifically we had:
gstav1parse.c[1850,11] in gst_av1_parse_detect_stream_format: Logic error: The left operand of '==' is a garbage value
gstav1parse.c[1606,11] in gst_av1_parse_handle_to_small_and_equal_align: Logic error: The left operand of '==' is a garbage value
Also a couple of false-positives:
gstav1parse.c[1398,24] in gst_av1_parse_handle_one_obu: Logic error: Branch condition evaluates to a garbage value
gstav1parse.c[1440,37] in gst_av1_parse_handle_one_obu: Logic error: The left operand of '-' is a garbage value
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2230>
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>
If a file includes a new version of a plugin that exits in the
registry, the output of gst-inspect is incorrect. The output has the
correct version but incorrect filename, and element description.
This seems to have also fixed some documentation issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1344>
Our decoder implementation does not use downstream d3d11 pool for
decoding because of special requirement of D3D11/DXVA. So preallocation
using the downstream buffer pool will waste GPU memory in most cases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2211>
This provides new HLS, DASH and MSS adaptive demuxer elements as a single plugin.
These elements offer many improvements over the legacy elements. They will only
work within a streams-aware context (`urisourcebin`, `uridecodebin3`,
`decodebin3`, `playbin3`, ...).
Stream selection and buffering is handled internally, this allows them to
directly manage the elementary streams and stream selection.
Authors:
* Edward Hervey <edward@centricular.com>
* Jan Schmidt <jan@centricular.com>
* Piotrek Brzeziński <piotr@centricular.com>
* Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2117>
This reverts commit 652773de36 and
modifies it to rename the caps field name to coded-picture-structure.
It was previously removed because it confuses the decoder and we didn't
have a valid use case for including it in the encoded caps at this
stage. We now do have such a use case but still don't want to confuse
the decoder, so the field is renamed.
However, it is still not accurate without looking at the SEI picture
structure of each frame, so it was named coded-picture-structure. If its
value is "frame" it is most likely progressive, if it's "field" it is
most likely interlaced or mixed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2177>
get_merged_collection() returns an owned stream collection and was
leaked in the else block.
Fix leak when running:
GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7,leaks:6" gst-play-1.0 --use-playbin3 test.mkv
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/954>
Make sure that the requested stream selection isn't identical to the current
one. If that's the case, just carry on as usual.
This avoids multiple `streams-selected` posting ... when the selection didn't
change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2185>
* glimagesink is not a recommended one on Windows
* Remove directdrawsink section
* d3dvideosink is legacy and should not be recommended
* Add d3d11videosink part
* directsoundsink should be deprecated
* Add wasapisink/wasapi2sink part
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2144>
The current way names the level by the number of B frames it contains, the
less it contains, the higher level it is. So the non ref B frames are in the
lowest layer and the B frames in the highest level refer to I/P frames.
But the widely used way is just the opposite, the ref B frames are in the
lower level and non ref B frames are at the highest level.
The is just a terminology change, and does not have any effect for compression
result and quality.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2149>
It doesn't matter for measurement purposes whether receiving them takes
a while and various PTP servers are not prioritizing to send them,
causing them to be dropped unnecessarily and preventing proper
synchronization with such servers.
This is especially a problem if the RTTs in the network are very low
compared to the additional delay imposed by the server.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2161>
timeapi.h is missing in our MinGW toolchain. Include mmsystem.h
header instead, which defines struct and APIs in case of our MinGW
toolchain. Note that in case of native Windows10 SDK (MSVC build),
mmsystem.h will include timeapi.h
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2153>
In case of re-syncing (i.e. moving to another partition to avoid too much of an
interleave), there was previously no checks to figure out whether a given
partition was already fully handled (i.e. when coming across it again after a
previous resync).
In order to handle this at least for single-track partitions, check whether we
have reached the essence track duration, and if so skip the partition.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2150>
The essence track position should only be overriden if we sucesfully switched to
another position. In case of EOS we do not want to override it else we would
increase the track position *again* at the end of this function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2150>
This field is used by DXVA/NVDEC/VA, and each specification
describes (NVDEC is not well documented) that it's the number of
bits used in short_term_ref_pic_set().
DXVA doesn't explicitly mention that whether the size of
emulation preventation bytes (EPB) is inclusive or not, but
VA is clearly specifying that it's the size after removing
EPB. Excluding EPB size here makes more sense therefore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1930>
The documentation could be read to mean that the caller continuous to
'own' the buffer, and that there is some other mechanism to find out
when to unref it.
Clarify that "not taking ownership" here means "taking a reference",
and specify that you can unref it at any time after calling the
function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2110>
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>
When building for Android, chances are that gstreamer is going to be
loaded from Java using System.loadLibrary(). In that case we can
initialize GStreamer (including static plugins), redirect log functions,
etc.
This code is copied from cerbero because it can be used with
gstreamer-full-1.0 too. Cerbero needs to be adapted to drop that code
and generate gst_init_static_plugins() function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/617>
Added GstVaFeature enum type, and new parameter for VA allocator's
set_format() and get_format(). Also added a new parameter in VA pool
gst_va_pool_new_with_config() and
gst_buffer_pool_config_set_va_allocation_params().
This new parameter will define if derived images will by used for
buffer mapping.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2057>
Expose description of vendor for user information, similar to
the description property of d3d11device.
Also, set description and DRM device path on GstContext structure
so that user can read them and it will be printed on terminal
when gst-launch-1.0 is used
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2064>
The console HANDLE will be keep signalled state unless application
reads console input buffer immediately. So we should read and flush
console input buffer from the thread where the event is signalled,
instead of GMain context thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2058>
Configure playsink tried element with the bus of the main pipeline.
That tried element can be a gl video sink, which would benefit from being
able to propagate context messages to the main pipeline and have other
internal pipeline elements configured with it. Having different elements
configured with the same GL context allows them to share buffers with
video/x-raw(memory:GLMemory) caps and achieving zero-copy.
Thanks to Alicia Boya García <aboya@igalia.com> for her work co-debugging
the issue and contributing to find a solution.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2056>
Sources that can internally handle buffering shouldn't have yet-another
buffering element after it. This can be simply detected by checking if it can
answer a TIME BUFFERING query just after creation.
If that is the case, we can expose the element source pads directly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>
By default, the classification is
"Converter/Filter/Colorspace/Scaler/Video/Hardware", but if VA
post-processor driver supports either color balance, skin tone
enhancement, sharpening or noise reduction, "Effect" is added.
Thus, if vapostproc ranking is raised, it can be chosen by
autovideosink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2066>
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>
If things progress fast enough, some state changes may not be seen be
the waiting code.
Fix by:
1. keeping a list of all the state changes
2. waiting checks each entry and if the relevant state is found, all
states up to and including then are removed.
This ensures that any waits will see all the state sets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
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>
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>
When creating a transceiver when creating an answer, the media kind of the
transceiver was never set correctly initially. This would lead to a
GST_WARNING being produced about changin a transceiver's media kind.
Fix by retrieving the GstSDPMedia kind from the offer instead as the answer
GstSDPMedia has not been set as the answer caps have not been chosen yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
In order to other plugins use gstva objects, such as allocators and buffer
pools, this merge request move them from the va plugin to the gstva library.
This objects are not exposed in <gst/va/gstva.h> since they are not expected
to be used by users, only by plugin implementators.
Because of the surface copy design, which is used to implement allocator's
mem_copy() virtual function, depends on the vafilter, which is kept inside
the plugin, memory copy through VAPosproc is disabled and removed temporarly.
Also added some missing parameter validation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2048>
Untabifying header file.
The logging category was moved from the plugin generic category to
the display category. It can argue that video formats hacks are
display dependant.
Added validations for input parameters.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2048>
... and add more encoding options.
QSV API supports dynamic bitrate change without IDR insertion.
That's more efficient way of runtime encoding option update
than starting from new sequence with IDR per bitrate option change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2039>
FFMPEG 5+ doesn't allow overriding the codec anymore (causes a segfault if you
attempt to do that). But the best part is ... that with the current caps
implementation in pad template and gst_ffmpeg_caps_to_codecid() we would never
replace it by anything different than the existing codec id.
Fixes#1054
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2052>
../gst-libs/gst/mpegts/gst-dvb-section.c:206:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
guint i = 0, allocated_events = 12;
^
../gst-libs/gst/mpegts/gst-dvb-section.c:365:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
guint i = 0, allocated_streams = 12;
^
../gst-libs/gst/mpegts/gst-dvb-section.c:543:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
guint i = 0, allocated_streams = 12;
^
../gst-libs/gst/mpegts/gst-dvb-section.c:885:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
guint i = 0, allocated_services = 8;
^
../gst-libs/gst/mpegts/gst-dvb-section.c:1316:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
guint i = 0, allocated_services = 8;
^
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
Check that `self` and `self->callback` are defined. `self` can be set to
`NULL` in `remove_listener`, and `self->callback` can be set to `NULL`
inside `gst_amc_surface_texture_jni_set_on_frame_available_callback`.
This can cause a segfault since the Java object can outlive the C
object, and call the callback after `remove_listener` is called.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2024>
Use the return value from gst_element_link_pads() and gst_bin_add()
Fixes:
../ext/gl/gstglmixerbin.c:305:12: error: variable 'res' set but not used [-Werror,-Wunused-but-set-variable]
gboolean res = TRUE;
^
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2038>
Produce an error if we try to use the feature of holding capture buffer
but it is not supported by the driver. Ingoring this can lead to stalls
as the driver will run-out of capture buffer to decode into. This
affects slice decoders but also split-field interlaced decoding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2009>
This flag is set when the stream is interlaced and the specific
slice is made of single parity fields rather the paired at the
macroblock layer. This is rarely needed in late decoding process
but the Rockchip RKVDEC HW interface requires it, hence needs to
be passed through V4L2 Stateless interface.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2009>
The official releases of libxml2 have been migrated to gitlab where
they are published for download via HTTP instead of FTP. Besides
adapting to the new location we now also get the benefit that the
tarball can be downloaded in restricted networks where FTP might be
blocked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2020>
Some problematic H265 stream may miss the reference frame in the DPB,
and get some message like: "No short term reference picture for xxx".
So there may be empty entries in ref_pic_list0/1 when passing to
decode_slice() function of sub class. We need to check the NULL pointer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2018>
Specify modules to look for OpenEXR when CMake is used, as we may have
CMake config files instead of pkg-config files that result from building
OpenEXR, which may be built with CMake which is typically the case on Visual
Studio builds.
In this case, Meson does seem to find the 'OpenEXR' package with CMake
after trying pkg-config, but does not consider it enough without the
'modules:' argument.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2014>
This reverts commit 3cad3455377d5a22faa138d9df840257059776c8.
That commit was breaking the association between an audio and
a video track in the standard case.
In practice, to support carrying separate MediaStream, we are
going a way to map what MediaStreamTrack belong to what MediaStream,
but that will require some thinking about the API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2023>
From https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-msid-16:
> Multiple media descriptions with the same value for msid-id and
> msid-appdata are not permitted.
Our previous implementation of simply using the CNAME as the msid
identifier and the name of the transceiver as the msid appdata was
misguided and incorrect, and created issues when bundling multiple
video streams together: the ontrack event was emitted with the same
streams for the two bundled medias, at least in Firefox.
Instead, use the transceiver name as the identifier, and expose
a msid-appdata property on transceivers to allow for further
customization by the application. When the property is not set,
msid-appdata can be left empty as it is specified as optional.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2003>
WebKit is not going to render anything until a URI is set, leading to a
WPE posting a `WPE View did not render a buffer` error message. To avoid
requiring the user to know it if they only want to use
`wpesrc::load-bytes` we can just use `about:blank` as default and
everything will work as users would expect.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1492>
This was not a problem here because even if we end up accidentally
linking to the wrong pad, things will work out eventually as long as
one pad-added is emitted for each pad that is added.
But it will be a huge problem if someone copies this code and changes
something that requires different handling for different sorts of
pads. The resultant code will be racy. Let's not do this, it's a bad
example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2008>
Adding new encoder elements nvd3d11{h264,h265}enc for Direct3D11
input support and re-written nvcuda{h264,h265}enc elements.
Newly writeen elements have some differences compared with old
nv{h264,h265}enc including non-backward compatible changes.
* RGBA is not a supported input format any more:
New elements will support only YUV formats to avoid implicit conversion
done by hardware. Ideally it should be done by upstream element
in order to have more control on it. Moreover, RGBA support can cause
redundant RGBA -> YUV conversion if multiple encoders are
used for the same RGBA input
* Subsampled planar format support is dropped:
I420 and YV12 format are not supported formats for Direct3D11.
Although it's supported in CUDA mode, it's not a hardware friendly
memory layout and it will waste GPU memory since UV planes
will have large padding due to the memory layout requirement of NVENC.
* GL support is dropped: Similar to the RGBA case,
GL support in encoder would be suboptimal if GL input is
used by multiple encoders, because each encoder will copy GL memory
into CUDA memory.
Upstream cudaupload element can be used for GL <-> CUDA
interop instead.
* No more pre-allocation of encoder input surfaces. New implementation
will use input CUDA memory without copy (zero-copy) or
will copy into a NVENC's input buffer struct in case of
system memory input.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1997>
Dispatches a list of active touch events to the wpe view on each
received TOUCH_FRAME event. Touch inputs currently only move the cursor,
since wpe doesn't seem to support clicking/scrolling or zooming with
touch input.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
Represents touchscreen events as a trail of black squares, one for each
reported position. Additionally, this adds the `display-mouse` and
`display-touch` properties to toggle visibility of mouse/touchscreen
events, since touchscreens often emulate mouse events, as well as
logging for all received navigation events.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
Add 5 new navigation event types for touchscreen events, with the same
naming and meaning as in libinput - touch-down, touch-motion, touch-up,
touch-frame and touch-cancel - as well as constructors and parse
functions for them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
Add a function to get x/y coordinates from suitable navigation events,
and one to create a copy with given coordinate values.
For e.g. translating event coordinates, this avoids having to either
switch on the event type to select the right parse function, or
having to rely on implementation details of the underlying event
structure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
This deprecates the current send_event interface, and the wrapper
functions based on it, replacing it with a send_event_simple interface and
wrapper function. Together with the new event constructors, this avoids
implementations having to directly access the underlying structure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
Since the strings are empty for GST_MSDK_CAPS_MAKE_WITH_DMABUF_FEATURE
and GST_MSDK_CAPS_MAKE_WITH_VA_FEATURE, when excuting
gst-inspect-1.0.exe msdkh265enc, there will be convert static caps error
because of the extra semicolon between two empty strings. Now macro
definitions are added to avoid this issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2004>
Pass the current frame to the duplicate_picture callback. This makes it easier
to set the frame's output_buffer if we already have one available. Also
documented that unlike VP9, it is not optional to implement this as the
picture will populate the DPB if it is a key-frame. To ensure this, remove the
default implementation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1992>
The system_frame_number is notably used by V4L2 decoder as a unique
indentifier for the frame that was decoded. This value is used to tell driver
which frame to reference, as V4L2 does not have an efficient mechanism to
otherwise pass back the frames.
For this reason, and because it is more ligical, copy the original
system_frame_number into the duplicate picture instead of using the current
frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1992>
Two RTP Header extensions are very relevant for rtprtxsend/receive.
1. "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id": will always be removed
2. "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id": will be written
instead of the "rtp-stream-id" header extension.
Currently it's only a simple replacement of one header extension for
another however a future change would only add the relevant extension
based on some heuristics (like, video frames only on one of the rtp key
frame buffers, or only until the rtx ssrc has been validated by the peer)
in order to reduce the required bandwidth.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1759>
Showing existing keyframe have special meaning in AV1. All the references
frame will be refreshed with the original keyframe information. The refresh
process (7.20) is implemented by saving data from the frame_header into the
state. To fix this special case, load all the relevant information into the
frame_header.
As there is nothing happening in between this and the loading of the key-frame
into the state, this patch also remove the separate API function, using it
internally instead.
Fixes#1090
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1971>
We need to parse the payload type map provided by the offer SDP and
set those values on the payloader, otherwise webrtcbin will create
a recvonly answer SDP and we won't send anything to the browser.
Fixed it for both C and Python sendrecv examples.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
Earlier, the example only supported one negotiation mode:
* Browser client is running, gstreamer starts a call and sends offer
Now these three modes are also supported:
* Browser client is running, gstreamer starts a call and sends an
offer request
* gstreamer connects and waits for browser client to start a call and
send an offer
* gstreamer connects and waits for browser client to start a call and
send an offer request
The following features are still missing:
* Data channel support
* TWCC support + stats logging
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
The documentation for several gst_*_writable_structure functions stated
that they would never return NULL, without making clear that the passed
object is required to be writable. This changes the wording in those
cases to make that requirement more clear.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1784>
When syncing to an RFC7273 clock this will add the original
reconstructed reference clock timestamp to buffers in form
of a GstReferenceTimestampMeta.
This is useful when we want to process or analyse data based
on the original timestamps untainted by any local adjustments,
for example reconstruct AES67 audio streams with sample accuracy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
When syncing to an RFC7273 clock this will add the original
reconstructed reference clock timestamp to buffers in form
of a GstReferenceTimestampMeta.
This is useful when we want to process or analyse data based
on the original timestamps untainted by any local adjustments,
for example reconstruct AES67 audio streams with sample accuracy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
When syncing to an RFC7273 clock this will add the original
reconstructed reference clock timestamp to buffers in form
of a GstReferenceTimestampMeta.
This is useful when we want to process or analyse data based
on the original timestamps untainted by any local adjustments,
for example reconstruct AES67 audio streams with sample accuracy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
We bind transceivers' fec_percentage property to the FEC encoder
percentage property, and with the binding bidirectional a deadlock
was introduced by the latest changes from !1762:
We take hold of the transceiver's object lock, then add the binding
and set the property to its initial value on the encoder, which causes
set_property to deadlock in the transceiver when the binding kicks in.
Changing the binding type to DEFAULT (source to target) is enough
to address the deadlock and still serves the original intent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1967>
Previously the result of the calculations included inaccuracies caused
by the NTP clock estimation, which caused the timestamps to jitter
+/- 1/clockrate.
By reorganizing the calculations it is possible to get rid of this
inaccuracy and calculate deterministic and exact packet timestamps based
on the actual NTP clock as long as the estimation is not off by more
than 2**31 clockrate units.
The only remaining inaccuracy that is introduced now is caused by the
conversion from the NTP clock to the pipeline clock.
Also split up debug output, demote many messages to the trace debug
level and output more intermediate results.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1955>
This is difficult to encounter in ordinary networks, but is
encountered when using tc-netem to add random delays to packets, and
also when your UDP stream is bonded over multiple links with varying
characteristics.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1952>
Holding previously decoded but not outputted pictures even after
new_sequence is not a safe approach in various aspect.
However, we cannot drain out DPB on new_sequence() unconditionally,
because there is a case where decoder should drop decoded pictures
if NoOutputOfPriorPicsFlag is set.
To detect NoOutputOfPriorPicsFlag before the new_sequence() call,
this patch splits decoding process into two path, one for nal unit parsing
in order to detect NoOutputOfPriorPicsFlag and then each nal unit
will be decoded.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1937>
Gapless playback is handled by adjusting buffer timestamps & durations
and by adding GstAudioClippingMeta.
Support for "Frankenstein" streams (= poorly stitched together streams)
is also added, so that gapless playback support doesn't prevent those
from being properly played.
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
1. Always set the according GstVaH264EncFrame pointer when GstVideoCodecFrame
pointer is assigned, which can make the logic safe.
2. Fix the forgotten change in _sort_by_frame_num. Its input pointer now is
GstVideoCodecFrame type.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1935>
Add properties to control input cropping in the V4L2 device.
The input cropping is applied before composing the result to the
capture buffer. By default the capture size will be set to the same
size as the crop region, but it can be scaled to a different output
frame size if supported by the V4L2 device.
If scaling is not supported, the cropped image will
be composed as is into the top-left corner of the capture buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1089>
Get the current crop bounding region from the V4L2 device so
that it can be provided to applications and used to validate
crop settings. Also make the default crop region available so
that it can be used to reset the crop when appropriate.
Uses the selection API when available with fallback to the crop
API for older kernels.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1089>
The gst_v4l2_object_set_crop() is used for removing buffer
alignment padding. Give it a name that better reflects
that usage. This helps to distinguish from cropping of the
input image (e.g. cropping at the image sensor on a captre
device), which can be unrelated to the memory buffer padding,
especially if scaling is involved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1089>
The default query handler would go through typefind, which by default accepts
any CAPS. But once configured, parsebin can't reconfigure itself, it should
therefore pass through the ACCEPT_CAPS query to the first element after
typefind (if any).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1900>
Don't reconfigure outputs when the select-streams
event is sent from the app, as the selection may
not take effect for some time. Instead, wait
for the pipeline to confirm the new set of
selected streams when it sends the message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1900>
If we previously had subtitles coming in, the video
may be chained through a text overlay block. Before,
the code would end up trying to link pads that were
already linked and video would not get reconnected
properly.
To fix that, make sure that the candidate
pads are actually unlinked first. If a textoverlay
is present and no longer needed, it will be cleaned
up later in the reconfiguration sequence.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1900>
Requesting a new pad can start a reconfiguration cycle, where
playsink will block all input pads and wait for data on them
before doing internal reconfiguration. If a pad is released,
that reconfiguration might never trigger because it's now waiting
for a pad that doesn't exist any more.
In that case, complete the reconfiguration on pad release.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1180>
Make it posible to configure the element to obtain the timestamps from
reference timestamp meta data instead of using the ntp-offset property,
or estimating its own offset. Currently the only time format supported
is "timestamp/x-unix", i.e. UTC time expressed in the unix time epoch.
In addition the custom event GstNtpOffset has been renamed to
GstOnvifTimestamp, to reflect that it is not necessarily used to convey
the ntp-offset. As a consequence we had to modify a couple of files in
the rtsp-server as well.
Fixes#984
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1683>
This patch fixes a seg.fault in gst_structure_new() with warnings as below.
GLib-GObject-WARNING **:
../gobject/gtype.c:4330: type id '0' is invalid
GLib-GObject-WARNING **:
can't peek value table for type '<invalid>' which is not currently referenced
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1918>
On GstVideoDecoder::{drain,flush}, we send null packet with
CUVID_PKT_ENDOFSTREAM flag to drain out decoder. Which will
reset CUVID parser as well.
To continue decoding after the drain, the next input buffer
should include sequence headers otherwise CUVID parser will
not report any decodeable frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1911>
There could be a case where the new program has the same program number as the
previous one ... but is actually located on a PID previously used for elementary
stream. In that case the program is guaranteed to not be an update of the
previous program but a completely new one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1893>
We need to be able to look for programs by their PID also. Using a hash table
was a bit sub-par (and overkill) for storing a range of programs.
This is needed because there could potentially be two programs with the same
program id but different PMT PID (while one is being deactivated the new one
would "exist").
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1893>
This commit modifies the interleave calculation to allow growing when incoming
data is before the segment start.
The rationale is that there is no requirement whatsoever for data before the
segment start to be "coherent" on all streams.
For example, a demuxer could rightfully send data from the video stream from the
previous keyframe (potentially quite a bit before the segment start) and the
audio from just before the segment start.
This will activate the same logic as growing the interleave when some streams
haven't received buffers yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1892>
* When a stream receives EOS, it will no longer change, we shouldn't take that
stream into account for interleave calculation.
* When streams (re)appear, we do not want to grow the initial interleave values
to excessive values. Instead of setting it to a default of 5s, progressively
grow it to that maximum.
* When the status of input streams change (i.e. going to/from "some haven't
received data yet" and "all have received data"), update the interleave
immediately instead of waiting for (potentially) 5s of data before updating
it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1892>
The point here is that rtpsession will create a new rtpsource when
the field "rtx-ssrc" is present, and when not doing rtx, that means
a random ssrc will create a new rtpsource that will be included in RTCP
messages for the current session.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1882>
Instead of using GstMiniObject to hold H264 frame, now it uses a plain
structure. Besides, instead of holding a reference to
GstVideoCodecFrame, the H264 frame structure is set as a
GstVideoCodecFrame user data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1856>
According to va_dec_hevc.h, pic_param->st_rps_bits should be set
for accelorater to skip parsing the *short_term_ref_pic_set
(num_short_term_ref_pic_sets) structure.
Also modified fill_picture to get parser info as a parameter,
in order to get slide_hdr->short_term_ref_pic_set_size.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1886>
Fix a small race where a group can receive stream-start
and post a pending buffering message just as another
thread posts a different buffering message, causing them
to be received by the application out of order. In the
worst case, this leads the application receiving a
stale 99% buffering message and going back to buffering
right after the 100% buffering message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1840>
Apparently GtkSharp expects each object has only one ToggleRef at any
time. Assigning element.Handle into Raw has a consequence that second
ToggleRef attempts to get created but fails on g_object_unref () that
breaks a GObject assertion:
toggle_refs_notify: assertion failed: (tstack.n_toggle_refs == 1)
This is because toggle references should be removed with
g_object_remove_toggle_ref(), not a simple unref().
In order to avoid duplicate toggle references, introduce
ElementFactory.MakeRaw(), which creates a GstElement without its
accompanying C# object. The returned raw pointer can be assigned into
another GLib.Object without trouble.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1885>
And use the output segment position for the outgoing timestamp while it
is. This is needed to delay the calculation of `output_ts_offset` until
we actually have a usable timestamp, as tsmux will output a few initial
packets while `last_ts` is still unset.
Without this, the calculation would use the initial `0` value, which did
not have the intended effect of making VBR mode behave like CBR mode,
but always calculated an offset equal to the selected start time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1884>
Fix suppression to support release and debug builds.
Here is the debug build call stack:
```
==10707== by 0x48B5520: g_malloc (gmem.c:106)
==10707== by 0x48D19DC: g_slice_alloc (gslice.c:1069)
==10707== by 0x48D3947: g_slist_copy_deep (gslist.c:619)
==10707== by 0x48D38B8: g_slist_copy (gslist.c:567)
==10707== by 0x4ADC90B: gst_debug_remove_with_compare_func (gstinfo.c:1504)
```
In release build `g_slist_copy (gslist.c:567)` got inlined:
```
==15419== by 0x48963E0: g_malloc (gmem.c:106)
==15419== by 0x48AA382: g_slice_alloc (gslice.c:1069)
==15419== by 0x48AB732: g_slist_copy_deep (gslist.c:619)
==15419== by 0x4A39B8F: gst_debug_remove_with_compare_func (gstinfo.c:1504)
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1814>
When doing only a single stream of audio/video this hardly matters,
but when doing many at the same time, the fact that you have to get
a hold of the glib global type-system lock every time you process a buffer,
means that there is a limit to how many streams you can process in
parallel.
Luckily the fix is very simple, by doing a cast rather than a full
type-check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1873>
There is a chance that pool->buffers[index] sets BUFFER_STATE_QUEUED, but
it has not been queued yet which makes pool->buffers[index] still NULL.
At this time, if pool_streamff release all buffers with BUFFER_STATE_QUEUED
state regardless of whether the buffer is NULL or not, it will cause segfault.
To fix this, also check buffer when streamoff release buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1842>
When tunneling over HTTP, if connection on the second channel happens
before the control timer is created we may trigger an assert in
rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
attaching the client thread to the context.
Fixes#1025
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1867>
When there is vpp scaling downstream, we need to make sure SFC is not
triggered because vpp may fall into passthrough mode which causes
the decoder negotiation to create src caps with vpp scaled width/height.
This patch includes bitstream's original size in first query with
downstream in gst_msdkdec_src_caps, which is the same for what we do for
color format in this query. This is to ensure SFC scaling starts to
work only when downstream directly asks for a different size instead of
through vpp.
Note that here SFC scaling follows the same behavior as msdkvpp:
if user only changes width or height, e.g. dec ! video/x-raw,width=xx !,
the height will be modified to the value which fits the original DAR.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1838>
* Hide GstCudaMemory member variables
* Make GstCudaAllocator object GstCudaContext independent
* Set offset/stride of memory correctly via video meta
* Drop GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT support.
This implementation actually does not support custom alignment
because we allocate device memory via cuMemAllocPitch
of which alignment is almost uncontrollable
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1834>
* cudaupload/download
- Specify only formats actually we can deal with
nvcodec elements, not all video formats
- Supports CUDA output for download and input for upload in order
to make passthrough possible, like other upload/download elements.
* cudabasetransform
- Reset conversion element if upstream CUDA memory
holds different CUDA context and the element can accept it.
This is the same behavior as corresponding d3d11 filter elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1834>
This was to support very old V4L2 kernel. As we moved to DMABuf and can now
detach buffers on renegotiation, the buffer it tries to fix no longer exist.
The risk to blocking indefinitly the application does still exist though.
Fixes#1070
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1861>
When we negotiate with downstream, We should use the intersected
caps of input and output to decide the alignment and stream format.
The current code just uses the input caps which may lack the stream
format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1837>
The demux now outputs the AV1 stream in "tu" alignment, so we do not need
to detect the input alignment. But the annex b stream format is not recognized
by the demux, we still need to detect that stream format for the first input.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1837>
Decoders that required frame aligmment and didn't have an associated
alpha decoder were skipped. This is because the parser was constructing
caps based on the software alpha decoder, which specify super-frame
alignment.
Iterate over the caps to filter the one that have a matching codec-alpha, with
the semantic the no codec-alpha field means codec-alpha=false. Then if
everything was removed, callback to the original, so that the first non-alpha
decoder will be picked.
Fixes#820
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1855>
Currently for copying the coded buffer onto a GStreamer buffer, the
coded buffer is mapped two times: one for getting the size, and later
for do the actual copy. We can avoid this by doing directly in the
element rather than in the general encoder object.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1845>
We need to always add the RTX/RED/ULPFEC elements as rtpbin will only
call us once to request aux/fec senders/receivers.
We also need to regenerate the media section of the SDP instead of
blindly copying from the previous offer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1762>
When the size of V4L2 capture or output is changes with VIDIOC_S_FMT,
the device is only required to update the compisition window to fit
inside the new frame size. This can result in captured data only being
updated on a portion of the frame after a resize.
Update the composition window to the default value determined by the
V4L2 device driver whenever the format is changed to make sure that
all image data is composed to its full size.
Fixes#765
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1806>
As specified formally in RFC8851
Each rid description is placed in its own caps field in the structure.
This is very similar to the already existing extmap-$id sdp<->caps
transformations that already exists.
The mapping is as follows:
a=rid:0 direction ';'-separated params
where direction is either 'send' or 'recv'
gets put into a caps structure like so:
rid-0=(string)<"direction","param1","param2",etc>
If there are no rid parameters then the caps structure is generated to
only contain the direction as a single string like:
rid-0=(string)direction
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1760>
Not having this field is equivalent with it being 1/1 so consider
it like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1826>
Not having these fields is equivalent with them being mono/0 so consider
them like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1826>
Otherwise fetching of the offer will fail with a cryptic error:
```
Traceback (most recent call last):
File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 56, in on_offer_created
offer = reply['offer']
TypeError: 'Structure' object is not subscriptable
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
```
ERROR peer '5762' not found
Traceback (most recent call last):
File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 190, in <module>
res = loop.run_until_complete(c.loop())
File "/usr/lib64/python3.10/asyncio/base_events.py", line 641, in run_until_complete
return future.result()
File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 155, in loop
self.close_pipeline()
File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 142, in close_pipeline
self.pipe.set_state(Gst.State.NULL)
AttributeError: 'NoneType' object has no attribute 'set_state'
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
```
File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 189, in <module>
loop.run_until_complete(c.connect())
File "/usr/lib64/python3.10/asyncio/base_events.py", line 641, in run_until_complete
return future.result()
File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 40, in connect
self.conn = await websockets.connect(self.server, ssl=sslctx)
File "/home/nirbheek/.local/lib/python3.10/site-packages/websockets/legacy/client.py", line 650, in __await_impl_timeout__
return await asyncio.wait_for(self.__await_impl__(), self.open_timeout)
File "/usr/lib64/python3.10/asyncio/tasks.py", line 445, in wait_for
return fut.result()
File "/home/nirbheek/.local/lib/python3.10/site-packages/websockets/legacy/client.py", line 654, in __await_impl__
transport, protocol = await self._create_connection()
File "/usr/lib64/python3.10/asyncio/base_events.py", line 1080, in create_connection
transport, protocol = await self._create_connection_transport(
File "/usr/lib64/python3.10/asyncio/base_events.py", line 1110, in _create_connection_transport
await waiter
File "/usr/lib64/python3.10/asyncio/sslproto.py", line 631, in _on_handshake_complete
raise handshake_exc
File "/usr/lib64/python3.10/asyncio/sslproto.py", line 676, in _process_write_backlog
ssldata = self._sslpipe.do_handshake(
File "/usr/lib64/python3.10/asyncio/sslproto.py", line 116, in do_handshake
self._sslobj = self._context.wrap_bio(
File "/usr/lib64/python3.10/ssl.py", line 526, in wrap_bio
return self.sslobject_class._create(
File "/usr/lib64/python3.10/ssl.py", line 865, in _create
sslobj = context._wrap_bio(
ssl.SSLError: Cannot create a client socket with a PROTOCOL_TLS_SERVER context (_ssl.c:801)
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
asyncio.get_event_loop() will not implicitly create a new event loop
in a future version of Python, so we need to do that explicitly.
```
webrtc_sendrecv.py:188: DeprecationWarning: There is no current event loop
loop = asyncio.get_event_loop()
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
If the tracks element was parsed from the SeekEntry, don't
parse it a second time and recreate tracks, as this
loses any tags that were read using the seek table.
If a genuinely new Tracks element is found, do read that
as it is needed for MSE support.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1798>
Remove the symbolic link `gst-uninstalled` which points to `gst-env`.
The `uninstalled` is the old name and the project should stick to a
single name for the procedure.
Remove the term from all the files, exceptions are variables from
dependencies like `uninstalled_variables` from pkgconfig and
`meson-uninstalled`.
Adjust mentions of the script in the documentation and README.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
Do not maintain similar build instructions within each gst-plugins-*
subproject and the subproject/gstreamer subproject. Use the build
instructions from the mono-repository and link to them via hyperlink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
This a new VA-API implementation of a H264 encoder.
It can control the GOP and parameter settings, while the MV searching,
VCL and the rate control algorithm are implemented by VA drivers and HW.
It supports most of the common usage options in H264, but still lacks
of look ahead, field, B frame weighted prediction, etc.
Co-authored-by: Victor Jaquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
The g_queue_clear_full() and g_array_copy() functions in the glib
may not be available for the current glib version check, so we add
helper functions to wrap it.
This should be deleted after the glib version bumps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
LOAD macro relies in m7 being zero for interleaving purposes. Using LOAD
on the m7 register makes it interleave with its new content instead of
with 0.
The effect of this bug was bobbing on some static lines that appeared
over fast-moving content.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1816>
The fd was in different meanings on windows:
POSIX read and write use the fd as a file descriptor.
The gst_poll use the fd as a WSASocket.
This patch use WSASocket as default on windows. This is a temporary measure, because IPC has many different implement. There may be a better way in the future.
See #1044
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1791>
The dynamic resolution changes when
the sequence starts when the decoder detects a coded frame with one or
more of the following parameters different from those previously
established (and reflected by corresponding queries):
1.coded resolution (OUTPUT width and height),
2.visible resolution (selection rectangles),
3.the minimum number of buffers needed for decoding,
4.bit-depth of the bitstream has been changed.
Although gstreamer parser has parsed the stream resolution.
but there are some case that we need to handle resolution change event.
1. bit-depth is different from the negotiated format.
2. the capture buffer count can meet the demand
3. there are some hardware limitations that the decoded resolution may
be larger than the display size. For example, the stream size is
1920x1080, but some vpu may decode it to 1920x1088.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1381>
v4l2videodec do some refactoring so that it can support
dynamic resolution change event.
1.wrap the setup process of capture as a function,
as decoder need setup the capture again when
dynamic resolution change event is received.
2.move the function "remove_padding"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1381>
This allows downstream of a payloader to know the RTP header's marker
flag without first having to map the buffer and parse the RTP header.
Especially inside RTP header extension implementations this can be
useful to decide which packet corresponds to e.g. the last packet of a
video frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
osxaudiodeviceprovider now probes devices more than once to determine
if the device can function as both an input AND and output device.
Previously, if the device provider detected that a device had any output
capabilities, it was treated solely as an Audio/Sink. This causes issues
that have both input and output capabilities (for example, USB interfaces
for professional audio have both input and output channels). Such devices
are now listed as both an Audio/Sink as well as an Audio/Source.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1385>
The RTP payload seems to be required as it carries the frame count
information. Also, gst_rtp_base_payload_allocate_output_buffer had
the second argument incorrect.
Strangely some devices like Shanling MP4 and Sony XM3 would still
work without this while some like the Sony XM4 do not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
While this is slightly more expensive (~48% slower per random number) it
does not cause any measurable difference when running through a complete
audio conversion pipeline.
On the other hand its random numbers are of much higher quality and on
spectrograms for 32 bit to 24 bit conversion the difference is clearly
visible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1729>
The instant-rate value in the TrickMode enum is a
flag, but the other values are not. Move instant-rate
to the end of the enum and give it a value large enough
for it to be used without modifying the trick-mode
setting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1788>
Update x264enc long-name to be more than just the name. Then the
description also was updated to be longer than the long-name, and
similar to the plugin description.
Finally, as I am here, H264 was replaced by H.264 and x264 is only a
plugin (not plugins).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1771>
They can't be used in any useful way. The type of every GstMemory is
always GST_TYPE_MEMORY and the subtyping relationship has to be
implemented on top of that via the associated allocator and mem_type
string.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1764>
Scenario:
- Source 1 requesting and waiting a clock id
- Source 2 requesting and waiting on a clock id
- Test attempting to crank both sources in the same GstHarness
gst_test_clock_crank() originally dropped locks between the retrieving
of the next clock id and advancing to the next clock id. This would
mean that both sources would race each other attempting to complete
their clock waits. Sometimes the operations would be performed in the
correct order, other times they would not and a FALSE return value would
be produced.
This would lead to an assertion in gst_harness_push_from_src() expecting
that all clock cranks to succeed.
Fix by ensuring that the clock wait produced is dealt with before
processing the next by not dropping the relevant locks after retrieving
the next clock id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1299>
This is a minimal unit test the show that the stride extrapolation can work
with all pixel format we support. This minimal verify that the extrapolation
match the stride we set into GstVideoInfo with 320x240 for all the pixel
format we support. The tiles formats are skipped, since their stride is
set as two 16bit integers, and we also skip over palette planes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
As this element is single threaded, we only need to stop the objects to
allow changing the format again. Fixes assertion notably on shutdown and
on some other situation where the format may be set twice without
actually activating the element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
Many of the legacy APIs, specifically in the Linux Kernel, have a
single stride for the pictures. In this context, it is common
to extrapolate the other strides based on the selected pixel
format. Such function have been copy pasted from video4linux2
plugin into wayland, kms and v4l2codecs plugins.
This patch implements a generalized from of that function and
make it available to everyone through the video library.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
Unlike other simple tiled formats, the Mediatek HW use different tile size
per-plane. The tile size is scaled according to the subsampling. Effectively,
using the name 16L32S to represent linearly layout tiles of size 16x32 bytes
in the Y plane, and 16x16 in the UV plane. In order to make this specificity
discoverable, a new SUBTILES flags have been added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
If the video4linux device supports norms but has no norm set, norm is
returned as an uninitialized variable after the ioctl call, leading to
gst_v4l2_tuner_get_norm_by_std_id() returning a random norm from the
supported norms. Catch this case and instead return NULL to indicate
that no norm is setup.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1625>
We have the d3d11screencapturesrc element in d3d11 plugin
which is obviously better than this element in terms of performance
and design, so we don't need to make people be confused by two separate elements.
Let's pick the better implementation and remove unnecessary one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1750>
... instead of round(). Depending on framerate, calculated position
may not be clearly represented by using uint64, 30000/1001 for example.
Then the result of round() can be sliglhtly larger (1ns) than
buffer timestamp. And that will cause unnecessary frame delay.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1747>
It was assumed that the kernel parameters would match with the bitstream value
but instead the author when with another set of value. Surprisingly, this
makes no difference with the resulting fluster score.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1748>
If a serialized event arrives behind a buffer, it should not be send before
it. This fixes the pending event handling so that only early pending events,
the one that arrrived or was generated while the adapter was empty get send
before pushing buffer. All other events are not pushed after.
This issue lead the latency tracer to think our audio encoder did not have any
latency. This was testing with opusenc in a live pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1266>
Each stream may have its own segment timeline
(i.g., different segment.start or segment.base)
depending on edit-list and composition-to-decode atom.
Make sure whether time position of a stream has been actually
far behind than that of current target stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1352>
If present, add '-lsocket' and '-lnsl' to network_deps.
ext/curl/meson.build: add network_deps to dependencies
gst/festival/meson.build: same
sys/shm/meson.build: same
Fixes linking issues on Illumos distros.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1525>
Often, users will need to scale inputs (e.g.
with vaapipostproc) before they are submitted
to the vaapioverlay. However, this results in
multiple VPP passes/operations in the pipeline
which creates unnecessary process overhead.
This change allows for inputs to be submitted
at original scale to vaapioverlay with per-sinkpad
scale dimensions specified so they can be scaled
and blended/composited in a single VPP pass/operation
to avoid the unnecessary process overhead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1380>
Don't set VAAPI vpp blend flags if alpha == 1.0,
i.e. fully opaque. This can avoid extra processing
overhead on some drivers that apply blending
unconditionally when flags are present, even if the
end result is the same without blend flags (i.e. all
opaque alpha channels).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1380>
Note that AYUV and AYUV64 formats will be used to expand format
support, especially some packed YUV formats (e.g., Y410, YUY2)
are common DXGI formats used for hardware decoder/encoder on Windows
but those formats cannot be used as a render target. We need to handle
them differently without pixel shader help, using compute shader
for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1699>
The libjpeg-turbo internal state might not be correctly initialized for
the first frame in a stream, pull the frame stride from gstreamer frame
metadata instead, which is correct even for the first frame, and which
makes this code consistent with the surrounding lines.
Fixes: e6d83d8f96 ("jpegdec: Support libjpeg-turbo colorspace conversion")
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1687>
It is imperative that the libjpeg-turbo state is properly initialized
before jpeg_start_decompress() is called. Make sure cinfo.out_color_space
and cinfo.raw_data_out are set to their final values matching their peer
caps before calling jpeg_start_decompress().
Fixes: e6d83d8f96 ("jpegdec: Support libjpeg-turbo colorspace conversion")
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1687>
Pull out peer caps checking code into gst_jpeg_turbo_parse_ext_fmt_convert().
This code is used by libjpeg-turbo extras to determine whether peer is capable
of handling buffers into which libjpeg-turbo can directly decode data. This
kind of check must be performed before jpeg_start_decompress() is called in
gst_jpeg_dec_prepare_decode() as well as in gst_jpeg_dec_negotiate(), hence
the common code.
This commit does modify the code a little to make it easier to call from both
call sites without much duplication, hence the extra `if (*clrspc)` test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1687>
This reverts commit 2aa2477208.
The commit is completely wrong, libjpeg-turbo is perfectly capable
of decoding I420 (YUV) to RGB. The test case provided alongside the
aforementioned commit passes without this revert because it decodes
image of JCS_YCrCb color space, so the new `if (clrspc == JCS_RGB)`
condition is false on that image, and the libjpeg-turbo decoding
does not get used. The real bug is hidden by that commit.
The real problem is in the call order of gst_jpeg_dec_prepare_decode()
and gst_jpeg_dec_negotiate(). The gst_jpeg_dec_prepare_decode() calls
jpeg_start_decompress() which sets up internal state of the libjpeg,
however, neither cinfo.out_color_space nor cinfo.raw_data_out are
set correctly yet. Those two are set up in gst_jpeg_dec_negotiate()
which is called a bit later. Therefore, the real fix is the set up
cinfo.out_color_space and cinfo.raw_data_out before calling
jpeg_start_decompress(). This is however a separate patch.
Fixes: 2aa2477208 ("jpegdec: only allow conversions from RGB")
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1687>
Remove all the d3d11 and dxgi header version dependent ifdef
and bump the minimum requirement to d3d11_4.h and dxgi1_6.h.
We are already failing support old Visual Studio (Windows SDK actually)
such as Visual Studio 2015. Note that our MinGW toolchain satisfies
the requirement.
From runtime point of view, this change should be fine since
we are checking OS version with IUnknown::QueryInterface()
everywhere in order to check API availability
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1684>
We make all MSDK encoders declare "memory:VAMemory" feature. Then
the pipeline such as:
gst-launch-1.0 -vf filesrc location=xxx.h264 ! h264parse ! \
vah264dec ! msdkh265enc ! fakesink
will choose VA memory caps between the VA decoder and MSDK encoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1087>
The MSDK encoder's query function is not set and it just forwards
all query to its base class. We now need to answer the context
query correctly. Other VA plugins need to query the VA display.
By the way, the current query of "gst.msdk.Context" is also missing.
The other MSDK elements must depend on the bin's context message(
sent in context_propagate()) to set their MsdkContext correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1087>
The MSDK VPP's query function is not set and it just forwards
all query to its base class. We now need to answer the context
query correctly. Other VA plugins need to query the VA display.
By the way, the current query of "gst.msdk.Context" is also missing.
The other MSDK elements must depend on the bin's context message(
sent in context_propagate()) to set their MsdkContext correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1087>
The MSDK decoder's query function is not set and it just forwards
all query to its base class. We now need to answer the context
query correctly. Other VA plugins need to query the VA display.
By the way, the current query of "gst.msdk.Context" is also missing.
The other MSDK elements must depend on the bin's context message(
sent in context_propagate()) to set their MsdkContext correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1087>
We now can use the gst va lib's display to create our MSDK context,
and use its helper functions to simplify our code. The improved logic
is like this:
1. Every MSDK element should use gst_msdk_context_find() to find a MSDK
context from neighbour. If valid, reuse it.
2. Use gst_msdk_ensure_new_context(). It will first query neighbours
about the GstVaDisplay, if found(e.g. some VA element is connected),
use gst_msdk_context_from_external_display() to create a MSDK context.
3. Then, creating the MSDK context from scratch. It creates both the
display and MSDK context.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1087>
The VA display object from VA lib is a common defined object. which
contain the whole display things. It is easier to use, and more important,
we can share it with the other VA plugins and keep all the VA related
plugins working on the same GPU device.
We also delete the useless gst_msdk_context_get_fd() API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1087>
The current manner for deciding the new temporal unit is based on
temporal delimiter(TD) OBU. We only start a new temporal unit when
the TD comes.
But some streams do not have TD at all, which makes the output "TU"
alignment fail to work. We now add check based on the relationship
between the different layers and it can successfully judge the TU edge.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1634>
Some streams may have problematic OBUs at the beginning, which causes
the parse fail to detect the alignment and return error. For example,
there may be verbose OBUs before a valid sequence, which should be
discarded until we meet a valid sequence. We should let the parse
continue when we meet such cases, rather than just return error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1634>
Our D3D11/DXVA codecs implementation has been verified
during 1.18 and 1.20 development cycle and also via the Fluster
test framework. Similar to the case of nvdec and vtdec,
we can prefer hardware over software in most cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1672>
For artificial input (in unit tests), all six bytes of
constraint_indicator_flags in hevc_caps_get_mime_codec() can be
zero. Add a guard against an out-of-bounds error that occurred in that
case. Change variables to signed int so comparison with -1 works.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1677>
Duplicating a picture what was already a dup was leading to a crash. Rename
the custom picture flags as HOLDS_BUFFER to make its meaning clear. Then save
then ref and store the picture as userdata, so it can be obtained when
duplicating. Finally, mark the doplicated as HOLDS_BUFFER to avoid thinking it
holds a request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1681>
This patch adds a new parameter: hdr-tone-mapping (same as
vaapipostproc), if the HDR capabilites are availabe in driver, and
it's disabled by default.
If hdr-tone-mapping is enabled then HDR fields in sink caps are
processed in frames from HDR to SDR, removing those hdr fields in
source pad caps too.
hdr-tone-mapping is not enabled if a color conversion is also
requested, since it fails to process in the iHD driver, so far.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1258>
1. Use api_version variable rather than static string.
2. Remove pkgconfig generation since currently the library
is not installed, only used internally.
3. Rely on dependency "required" to abort compilation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1650>
In commit e699aaeb we moved linking of libgudev to the plugin rather
the library, because it's only used in the plugin. But the dependency
check is still done in library.
This patch removes the dependency check in library, and updates the
dependency check in plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1650>
Constantly updating the ts_offset results in audiable glitches
when streaming audio using ntp-sync=true. By requiring a minimum
offset before updating ts_offset this can be mitigated. Added a
parameter which can be used to set min_ts_offset in ntp-sync mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
A new implementation of Intel Quick Sync Video plugin.
This plugin supports both Windows and Linux but optimization for
VA/DMABuf is not implemented yet.
This new plugin has some notable differences compared with existing
MSDK plugin.
* Encoder will expose formats which can be natively supported
without internal conversion. This will make encoder
control/negotiation flow much simpler and cleaner than
that of MSDK plugin.
* This plugin includes QSV specific library loading helper,
called dispatcher, with QSV SDK headers as a part of this plugin.
So, there will be no more SDK version dependent #ifdef in the code
and also there will be no more build-time MSDK/oneVPL SDK
dependency.
* Memory allocator interop between GStreamer and QSV is re-designed
and decoupled. Instead of implementing QSV specific allocator/bufferpool,
this plugin will make use of generic GStreamer memory
allocator/bufferpool (e.g., GstD3D11Allocator and GstD3D11BufferPool).
Specifically, GstQsvAllocator object will help interop between
GstMemory and mfxFrameAllocator memory abstraction layers.
Note that because of the design decision, VA/DMABuf support is not made
as a part of this initial commit. We can add the optimization for Linux
later once GstVA library exposes allocator/bufferpool implementation as
an API like GstD3D11.
* Initial encoder implementation supports interop with GstD3D11
infrastructure, including zero-copy encoding with upstream D3D11 element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1408>
There's no need to do this, and it can make seeking far less accurate.
For a specific use case: I am working with a long (45-minute) MPEG-1 layer 3 file, which has a constant bit rate but no seeking tables. Trying to seek the pipeline immediately after pausing it, without the ACCURATE flag, to a location 41 minutes in, yields a location that is potentially over ten seconds ahead of where it should be. This patch improves that drastically.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/374>
During dispose the pool will still have a reference count of 1 and all
API on it can still be safely called.
Subclasses will have already freed their own data before finalize is
called but would nonetheless be called into again via the pool
deactivation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1645>
It's almost pointless and makes little sense as subclass might
want to modify refcount of the object or so. And all subclasses
are already casting them to non-const version as well.
In a general sense, we need to avoid passing refcounted object
with const qualifier.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1238>
... in order to make older g-i happy (~1.60) which doesn't like
freeform descriptions in the value_name field. Which in turn
then makes hotdoc happy instead of erroring out when we bump
the symbol index version.
We usually only (ab)use the name field for description strings
for private plugin enums, not for public API visible to bindings.
This lets glib-mkenum generate the _get_type() function for the
enum again, which in turn will generate the expected value names
to match the enums.
We might be able to add this back later once we can upgrade the
g-i version requirement (and the documentation job image).
This reverts most of commit b0aab48cdcf0a454d14aeb4d907209d8ee3f1add
There's a race condition in gsttagdemux.c between typefinding and the
end-of-stream event. If TYPE_FIND_MAX_SIZE is exceeded,
demux->priv->collect is set to NULL and an error is returned. However,
the end-of-stream event causes one last attempt at typefinding to occur.
This leads to gst_tag_demux_trim_buffer() being called with the NULL
demux->priv->collect buffer which it attempts to dereference, resulting
in a segfault.
The malicious MP3 can be created by:
printf "\x49\x44\x33\x04\x00\x00\x00\x00\x00\x00%s", \
"$(dd if=/dev/urandom bs=1K count=200)" > malicious.mp3
This creates a valid ID3 header which gets us as far as typefinding. The
crash can then be reproduced with the following pipeline:
gst-launch-1.0 -e filesrc location=malicious.mp3 ! queue ! decodebin ! audioconvert ! vorbisenc ! oggmux ! filesink location=malicious.ogg
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/967
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1620>
The kCVPixelFormatType_64RGBALE enum is only available on macOS Big
Sur (11.3) and newer. We also cannot use that while configuring the
encoder or decoder on older macOS.
Define the symbol unconditionally, but only use it when we're running
on Big Sur with __builtin_available().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1613>
Hotdoc should be able to extract and parse comments out of these. Just
need to be careful to only add the glob in directories that actually
contain *.m (objc) and *.mm (objcpp) files.
Also fix some doc comments and remove redundant ones.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1614>
Platform wise, is not possible, as far as I known, to have Wayland
without kernel's DRM. Though, it's possible to configure
gstreamer-vaapi without DRM but Wayland support, with the enhanced
handling of dmabuf in vaapisink for Wayland, vaapisink will always
fail. Given both issues, configuration with no DRM but Wayland, makes
things more complex, and a simpler approach is to refuse that
configuration.
This patch disables Wayland support if there isn't DRM support. Also,
it disables the display test for Wayland, relying only on DRM and
X11.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1606>
Formats map is instantiated at the end of the display
instantiation. The problem is the Wayland display which looks for a
format in a callback, before the map is populated.
If user compiles gstreamer-vaapi with DRM support, the map is
populated with a DRM display at GStreamer plugin registration. But if
not, or a VA driver is not available, the plugin will try with a
Wayland driver, which cause the NULL de-reference.
Nevertheless, in the case of no DRM support, and if the Wayland
display doesn't get a reply from the format conversion is not a
problem.
So the solution is the trivial one, check if the format map is already
populated before de-reference it.
Fixes: #977
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1606>
This is listed as a public interface implemented by osxaudio, so we
need to mark it as a plugin API so that it's listed in the
documentation correctly.
This is an ancient symbol, so add it to the symbol index too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1601>
These symbols from macOS plugins osxaudio, osxvideo, and applemedia
have been present for a very long time but were never documented.
This allows us to document these, and also add Since: markers for new
features (symbols) there were added in 1.20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1601>
This moves the ABI check to the registration, so we don't expose
decoders with the wrong ABI or that are just broken somehow. It
also makes few enhancement:
- Handle missing, but required controls
- Prints the controls macro name instead of id
This should fix RK3399 support with a currently release minor
regression in the Hantro driver that cause errors.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1599>
... in addition to all the other issues that were ignored for them
already.
At least the reverse playback test regularly times out, waiting for a
download to finish that has already finished successfully.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1593>
Starting with libsoup3, there is no attempt to handle thread safety
inside the library, and it was never considered fully safe before
either. Therefore, move all session handling into its own thread.
The libsoup thread has its own context and main loop. When some
request is made or a response needs to be read, an idle source
is created to issue that; the gstreamer thread issuing that waits
for that to be complete. There is a per-src condition variable to
deal with that.
Since the thread/loop needs to be longer-lived than the soup
session itself, a wrapper object is provided to contain them. The
soup session only has a single reference, owned by the wrapper
object.
It is no longer possible to force an external session, since this
does not seem to be used anywhere within gstreamer and would be
tricky to implement; this is because one would not have to provide
just a session, but also the complete thread arrangement made in
the same way as the system currently does internally, in order to
be safe.
Messages are still built gstreamer-side. It is safe to do so until
the message is sent on the session. Headers are also processed on
the gstreamer side, which should likewise be safe.
All requests as well as reads on the libsoup thread are issued
asynchronously. That allows libsoup to schedule things with as
little blocking as possible, and means that concurrent access
to the session is possible, when sharing the session.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/947
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1555>
Intel drivers expose some colorbalance's maximum values much more
bigger than their minimum values, given their middle values (default
value). This means, in practice, that the real middle point between
the maximum and minimum values implies a major change in the color
balance, which is not expected by the GStreamer color balance logic.
This patch makes the given maximum value symmetrical to the minimum
value, given the middle one (default value).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1580>
It was getting pulled in automatically via cairo, but the version
there is too old for us. We need the latest to fix Windows ARM64 NEON
support:
> ERROR: No specified compiler can handle file subprojects\libpng-1.6.37\arm/filter_neon.S
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1570>
As now, we warn if the decoder have no support src pixel format, but that
warning is called before the type (hence the debug category) is initialized.
Fix this by moving the debug category init out of the type initialization,
into the register funcitons.
This will fix an assertion that occures in the register function and allow
relevant log to be seen by the users.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1588>
Those will cause us to renegotiate at the next aggregate cycle,
and while at that point we may decide to reconfigure upstream
branches (in practice we don't as this is inherently racy,
and that's the reason why mixer subclasses perform conversion
internally), we certainly don't want to just forward the event
willy-nilly to all our sinkpads.
An actual issue this is fixing is when caps downstream of a
compositor are changed at every samples-selected signal emission,
for the purpose of interpolating the output geometry, and the
compositor has a non-zero latency, the reconfigure events were
forwarded to basesrc, which triggered an allocation query, which
in turn caused aggregator to have to drain (thus not being able
to queue <latency> frames), leading to disastrous effects
(choppy output as compositor couldn't consume frames fast enough,
the higher the latency the choppier the output)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1464>
The merge request workflow can be confusing to people unfamiliar with
it, so add screenshots.
Also add a new section on how to revise merge requests, since a lot of
people tend to open new merge requests to make any changes.
Eliminate the separate "How to Prepare a Merge Request for Submission"
section -- merge it with the main section.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1367>
This is usually necessary to allow gst-indent to treat it as
a statement, but we do not run gst-indent on headers and we do not
have extra semicolons in other places that this macro is used in the
header. Fixes warnings when using the header:
```
In file included from gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/video.h:185,
from XYZ:9001:
gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h:206:78: warning: ISO C does not allow extra ‘;’ outside of a function [-Wpedantic]
206 | G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstVideoAggregatorConvertPad, gst_object_unref);
| ^
gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h:214:181: warning: ISO C does not allow extra ‘;’ outside of a function [-Wpedantic]
214 | G_DECLARE_DERIVABLE_TYPE (GstVideoAggregatorParallelConvertPad, gst_video_aggregator_parallel_convert_pad, GST, VIDEO_AGGREGATOR_PARALLEL_CONVERT_PAD, GstVideoAggregatorConvertPad);
| ^
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1572>
When fixating color, there might be "other caps" with color spaces not
supported by the caps features exposed in the vapostproc's source pad
caps template (perhaps it's a bug somewhere else in GStreamer).
This solution checks if the proposed format exists in the filter
within the caps feature associated with the proposed format.
The check is done with the new filter's function
gst_va_filter_has_video_format().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1559>
In the need-data appsrc callback, a buffer is pulled from the
appsink. This buffer is then copied so that metadata is writable.
The copy is pushed to the appsrc but it doesn't take ownership
of the buffer so we need to manually unref it. The original buffer
is finally unreffed when the sample is freed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
This is due to an unsafe usage of the pad task. We didn't ensure proper
ownership of the task. That race involved the task being released too early,
and was detected, luckily, by the glib mutex implementationt that
reported the mutex being disposed while being locked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1478>
On Android (especially) and for static builds in general it is safer to link
against libsoup and have the dynamic custom loading disabled. For those cases we
can safely assume the application will use either libsoup2 or libsoup3 and not
both.
Fixes#939
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1536>
The earlier size of 2 MB was set back in 2009, it doesn't
seem unreasonable to raise it to 8 MB these days. The use
case at hand is matroskademux containing both a video stream
with a very low amount of compression but no decoding latency,
and a H265 stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1538>
It is an extremely common mistake on Windows to have incorrect PATH
values when loading a plugin, and the error from g_module_error()
(which just calls FormatMessageW()) is very confusing in this case:
The specified module could not be found.
https://docs.microsoft.com/en-us/windows/win32/debug/system-error-codes--0-499-#ERROR_MOD_NOT_FOUND
It implies the plugin itself could not be found. The actual issue is
that a DLL dependency could not be found. We need to detect this case
and print a more useful error message.
We should still print the error fetched from FormatMessage() so that
people are able to google for it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1540>
GstAudioRingBufferSpec::segsize has been configured by using
device period but GstWasapi2RingBuffer was referencing the
buffer size returned by IAudioClient::GetBufferSize()
which is most likely larger than device period.
Fixing to sync them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1533>
If the `area_surface` got unmapped when changing to the `READY` or
`NULL` state, we currently don't remap it when playback resumes and
`wp_viewporter` is supported. Without `wp_viewporter` we do remap
it, but rather unintentionally and also when not wanted.
On Weston this has not been a big problem as it so far wrongly maps
subsurfaces of unmapped surfaces anyway - i.e. only the black
background was missing on resume. On other compositors and future
Weston this prevents the `video_surface` to get remapped.
Shuffle things around to ensure `area_surface` is mapped in the
right situations and do some minor cleanup.
See also https://gitlab.freedesktop.org/wayland/weston/-/issues/426
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1483>
Latest ffmpeg has removed avcodec_get_context_defaults(), and its
documentation says a new AVCodecContext should be allocated for this
purpose. The pointer returned by avcodec_find_decoder() is now
const-qualified so we also need to adjust for it. And, AVCOL_RANGE_MPEG
is now rejected with strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1531>
The later, doing damage in surface coordinates instead of buffer
coordinates, has been deprecated. The reason for that is that it
is more prone to bugs, both on the client and the compositor side,
especially when paired with buffer scale, `wp_viewporter` or
buffer transforms.
Unfortunately, on Weston this risks running into
https://gitlab.freedesktop.org/wayland/weston/-/issues/446
(which causes trouble for several other projects as well). However,
that bug only affects cases where we run in sync mode, i.e. only
during resizes. In practise I haven't been able to observe the
issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
Each time we call `wl_surface_damage()` we want to do full surface
damage. Like Mesa, just use `G_MAXINT32` to ensure we always do
full damage, reducing the need to track the right dimensions.
`window->video_rectangle` is now unused, but we keep it around for
now as we may need it again in the future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
From the spec:
> This request is used to describe the regions where the pending
> buffer is different from the current surface contents
We currently also call `wl_surface_damage()` on surfaces without
new or still compositor-hold buffers, e.g. when resizing the window.
In that case we call it on `area_surface_wrapper`, even though it
gets resized via `wp_viewport_set_destination()`, in which case
the compositor is in charge of repainting the area on screen.
Doing so is currently not forbidden by the spec, however it might
be in the future, see
https://gitlab.freedesktop.org/wayland/wayland/-/issues/267
Thus lets stay close to the spec and only call `wl_surface_damage()`
when we just attached a buffer.
Right now this prevents runtime assertions in Mutter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
`gst_wl_window_set_opaque` does not get called on window resizes,
potentially leaving opaque regions too small.
According to the spec opaque regions can be bigger than the surface
size - parts that fall outside of the surface will get ignored.
Thus we can can simply use `G_MAXINT32` and be sure that the whole
surfaces will always be covered.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
If the VANC track does contain packets, but we skip over all packets, just
treat it the same as if there hadn't been any packets at all and send a
GAP event instead of erroring out with "Failed to handle essence element".
We would error out because when we reach the end of the loop without having
found a closed caption packet the flow return variable is still FLOW_ERROR
which is what it has been initialised to.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1518>
Though the profiles[0] is inited as GST_H265_PROFILE_INVALID in the
gst_h265_profile_tier_level_get_profile(), the profile detecting may
change its content later. So the return of profiles[0] may not be an
invalid profile even the len is 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1517>
The previous code was mistakenly trying to compute a cc_type out
of the first byte in the byte triplet, whereas it is to be interpreted
as:
> Bit b7 of the LINE value is the field number (0 for field 2; 1 for field 1).
> Bits b6 and b5 are 0. Bits b4-b0 form a 5-bit unsigned integer which
> represents the offset
The same mistake was made when creating padding packets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1496>
Sometimes we can't output anything because we don't have enough
incoming frames. In that case, the resampler was trying to call
do_quantize() and do_resample() in a loop forever because there would
never be samples to output (so chain->samples would always be NULL).
Fix this by not calling chain->make_func() in a loop -- seems
completely unnecessary since calling it over and over won't change
anything if the make_func() can't output samples.
Also add some checks for the input and / or output being NULL when
doing conversion or quantization. This will happen when we have
nothing to output.
We can't bail early, because we need resampler->samples_avail to be
updated in gst_audio_resampler_resample(), so we must call that and
no-op everything along the way.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1461>
When the image is opaque but the output ProRes format has an alpha
component (4 component, 32 bits per pixel), Apple requires that we
signal that it should be ignored by setting the depth to 24 bits per
pixel. Not doing so causes the encoded files to fail validation.
So we set that in the caps and qtmux sets the depth value in the
container, which will be read by demuxers so that decoders can skip
those bytes entirely. qtdemux does this, but vtdec does not use this
information at present.
The sister change was made in qtmux and qtdemux in:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1061
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1489>
Old "application/*" are now as per RFC8081 deprecated in favor of
new "font/*" mime types. Some new encoders are already using the
updated mime types. We need to also add them to the support list
in order for assrender to correctly identify them as fonts.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1481>
If there is no jitterbuffer stats we should not attempt to store them in the
global stats structure.
Also add a g_return_if_fail in _gst_structure_take_structure() about this
because it is a programmer error to pass an invalid pointer address there.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1479>
Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
will avoid the following build failure, because the correct girdir
location will be retrieved from gstreamer-1.0.pc:
/home/giuliobenetti/autobuild/run/instance-3/output-1/host/riscv32-buildroot-linux-gnu/sysroot/usr/bin/g-ir-compiler gst/rtsp-server/GstRtspServer-1.0.gir --output gst/rtsp-server/GstRtspServer-1.0.typelib --includedir=/usr/share/gir-1.0
Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
Fixes:
- http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
Prior to this patch, we considered that a stream was blocking
whenever a pad probe was triggered for either the RTP pad or
the RTCP pad.
This led to situations where we subsequently unblocked and expected
to find a segment on the RTP pad, which was racy.
Instead, we now only consider that the stream is blocking when
the pad probe for the RTP pad has triggered with a blockable object
(buffer, buffer list, gap event).
The RTCP pad is simply blocked without affecting the state of the
stream otherwise.
Fixes#929
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
Instead of a sequence of if statements, declare a table to map profile
idc with profiles and traverse it.
Also, first add the profile from the parsed profile idc and later add,
into the profile array, the profile from the compatibility flags.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1440>
It's possible a HEVC stream to have multiple profiles given the
compatibility bits. Instead of returning a single profile, internal
gst_h265_profile_tier_level_get_profiles() returns an array with all
it possible profiles.
Profiles are appended into the array only if the generated profile
is not invalid.
gst_h265_profile_tier_level_get_profile() is rewritten in terms of
gst_h265_profile_tier_level_get_profiles(), returning the first
profile found the array.
And gst_h265_get_profile_from_sps() is also rewritten in terms of
gst_h265_profile_tier_level_get_profiles(), but traversing the array
verifying if the proposed profile is actually valid by Annex A.3.x of
the specification.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1440>
BT.2020 color primaries are designed to cover much wider range of
CIE chromaticity than BT.709, and also it's used for both SDR and HDR
contents. So, the incorrect assumption (i.e., BT.709 as a BT.2020)
is risky and resulting image color tends to be visually very wrong.
Unless there's obvious clue, don't consider color space of high resolution
video stream as BT.2020
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1445>
* Add fec / red encoders as direct children of webrtcbin, instead
of providing them to rtpbin through the request-fec-encoder signal.
That is because they need to be placed before the rtpfunnel, which
is placed upstream of rtpbin.
* Update configuration of red decoders to set a list of RED payloads
on them, instead of setting the pt property.
That is because there may be one RED pt per media in the same session.
* Connect to request-fec-decoder-full instead of request-fec-decoder,
in order to instantiate FEC decoders according to the payload type
of the stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
When multiple streams are bundled together, there may be more
than one red payload type to handle.
In addition, as the red decoder works by filling in gaps in
the seqnums, there needs to be one rtp_history queue per sequence
domain.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
In redenc, when input buffers have a header for the TWCC extension,
we now add one to our wrapper buffers.
In ulpfecenc we add one in that case to our protection buffers.
This makes TWCC functional when UlpRed is used in webrtcbin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1414>
In order to allow "level-asymmetry-allowed" we now handle a new
"profile" field, which as the same semantics as the "profile" field in
H.264 stream so that we can force payloaded stream to have the right
format when using the `gst_sdp_media_get_caps_from_media` to set caps
filter after the payloader. This allows a simple negotiation in standard
RTP negotiation based on SDPs (like webrtc) for that particular case,
closely respecting the specs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>
The ["level-asymmetry-allowed"] field states that the peer wants the
profile specified in the "profile-level-id" fields but doesn't care
about the level. To express this in GStreamer caps term, we add a
"profile" field in the caps, which reuses the usual "profile" semantics
for H.264 streams and, and remove "profile-level-id" and
"level-asymmetry-allowed" fields.
["level-asymmetry-allowed"]: https://www.iana.org/assignments/media-types/video/H264
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>
We are querying supported swapchain colorspace via
CheckColorSpaceSupport() but it doesn't seem to be reliable.
Use only tested full-range RGB formats which are:
- sRGB
- BT709 primaries with linear RGB
- BT2020 primaries with PQ gamma
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1433>
When using playbin3, it seems that the alpha decode is always first to
push caps and run an allocation query. As the format change from sink
and alpha were not synchronized, the allocation query could endup
being run before the caps are pushed. That may lead to failing query,
which makes the decoder thinks there is no GstVideoMeta downstream and
most likely CPU copy the frame.
This patch implements a format cookie to track and synchronize the
format changes on both pads fixing the racy performance issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1439>
This adds the alignment field to the template caps. Without this field
set, the auto-plugger will see fixed caps and will use
gst_caps_is_subset() against the caps produced by the parser. This is a
challenge for all cases where a parser can do conversion. This is fixed
by adding alignment field, which makes the auto-pluggers do an
intersection of the caps as it gets unfixed caps after intersection now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1439>
XNextEvent() blocks indefinitely in absence of X11 events, which can
prevent the pipeline from stopping.
This can cause problems when ximagesrc is used in "remote desktop"
scenarios and the GStreamer application itself, through which the user
is viewing and controlling the machine, is the only source of input
events.
Replace the call with non-blocking XCheckTypedEvent().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1438>
since `gst_caps_replace()` and `gst_pad_set_caps()` both ref the caps and neither of them takes the ownership of the caps -> it must be unreffed in `gst_multi_file_src_set_property()`
to test the leak (on Unix): `echo coucou > /tmp/file.txt && GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7" gst-launch-1.0 multifilesrc location=/tmp/file.txt caps='txt' ! fakesink`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1436>
When detecting the remote time has been reset which may occur if remote
device providing the clock server has been power reset, then clock is
no longer synced. Setting clock state will trigger a signal to client
informing on sync lost making it possibility to take appropriate action.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/975>
Due to a copy paste bug, the bitdepth was never set and that was leading
to requesting sizeimage of 0. Previously that worked since the driver
would in that case pick a size for us. But now the we bumped the minimum
to 4KB, the driver happily allocate 4KB of bitstream which lead to
decoding error.
As MPEG2 have a fixed bitdeph of 8, use a define instead of the run-time
variable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1415>
vaapidecode is used in vaapidecodebin and it exposes all the
theoretically supported caps, but that slows down autoplug. With this
autplug is negotiated faster, giving more option to decodebin to select
other decoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1405>
The V4L2 uAPI uses pic_num for both PicNum and ShortTermPicNum. It also
doe the same for both FrameNum and LongTermFrameIdx. This change does
not change the fluster score, but fixed a visual corruption noticed
with some third party streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1387>
The current code does not set the copied memory correctly when it is popped
from the surface cache pool.
1. We forget to ref the allocator, which causes the allocator to be freed
unexpected, and we get a crash later because of the memory violation.
2. We forget to add ref_mems_count, which causes the surface leak because
the surface can not be pushed back to the cache pool again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1373>
Set minimum sizeimage such that there is enough space for any overhead
introduced by the codec.
Notably fix a vp9 issue in which a small image would not have a
bitstream buffer large enough to accomodate it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1012>
Rework gstvp9{decoder|statefulparser} to optionally parse compressed headers.
The information in these headers might be needed for accelerators
downstream, so optionally parse them if downstream requests it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1012>
Also ignore 0x0 sizes in the fallback case and assume the size can be
anything between 1x1 and MAXxMAX.
This fixes the case where a width=0, height=0 caps are created. Whith
this patch the caps will contain width=[1,MAX], height=[1,MAX].
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1396>
When an extmap is defined twice for the same ID, firefox complains and
errors out (chrome is smart enough to accept strict duplicates).
To work around this, we deduplicate extmap attributes, and also error
out when a different extmap is defined for the same ID.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1383>
There are a lot of info in the mpeg2's sequence(also including ext
display_ext and scalable_ext). We need to notify the subclass about
its change, but not all the changes should trigger a drain(), which
may change the output picture order. For example, the matrix changes
in sequence header does not change the decoder context and so no need
to trigger a drain().
Fixes: #899
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1375>
Follow-up on 97d83056b3, only check
for intersection with the current srccaps when checking if a sinkpad
can accept caps.
I must have been lucky in my firefox testing then, and always entered
the code path with audio getting negotiated first, thus not failing
the is_subset check when srccaps had been negotiated as
application/x-rtp, and an accept-caps query was made for the video
caps with a defined extmap.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1384>
This is a requirement for GstPlayer when using the default overlay interface
provided by the pipeline. The GstPlayerWrappedVideoRenderer requires a valid
pipeline, but that's available only after the GstPlay thread has successfully
started.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1345>
Dynamically registered elements (hardware element in most cases)
may or may not be available on a system and properties may be different
per system.
This new API will make documentation skipping possible in programmable way.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1360>
ShowWindow() could be blocked while doing gst_d3d11_window_win32_unprepare
when external window handle provided to d3d11videosink in multi-threaded
environment.
The condition that issue happened is, UI thread is waiting for a
background thread that changes d3d11videosink state to NULL, and the
background thread would try to send a window message to the queue.
The queue is already occupied by the UI thread, so the background
thread will be blocked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1366>
Instead of defining a sized array for function signature, use it
unsized (a pointer alias, basically). In this way clang warning is
silenced:
warning: ‘fill_profiles’ accessing 64 bytes in a region of size 12 [-Wstringop-overflow=]
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1357>
If qt asks us to redraw before we have both set a buffer and caps we
would attempt to use the new caps with the old buffer which could result
in bad things happening.
Only update caps from new_caps once the buffer has actually been set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1346>
Make sure the EGLImage we're rendering to the GL memory stays alive long enough,
until the the GL memory has been destroyed.
This change fixes tearing and black flashes artefacts that were happening
because the EGLImage was sometimes destroyed before the sink actually rendered
the associated texture.
Fixes#889
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1354>
This takes a plain message string and not a format string, and as a
result doesn't have to be passed through vasprintf() and lead to further
unnecessary allocations. It can also contain literal `%` because of
that.
The new function is mostly useful for bindings that would have to pass a
full string to GStreamer anyway and would do formatting themselves with
language-specific functionality.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1356>
There's a potential race condition with this sort of pipelines on
certain systems (depends on the processing load):
GST_DEBUG_DUMP_DOT_DIR=/tmp \
gst-launch-1.0 uridecodebin3 uri=file://stream.mp4 ! glupload ! \
glimagesink --gst-debug=*:4
Right after the pipeline passes from PAUSED to READY, bin_to_dot_file
dumps uridecodebin3 properties, but current uri and suburi might be
already freed, causing a potential use-after-freed.
This patch makes NULL the current item right after all the play items
are freed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1353>
A previous patch has caused rtpfunnel to output twcc-related
information downstream, however this leaked into upstream
negotiation (through funnel->srccaps), causing payloader to
negotiate twcc caps even when not prompted to do so by the user.
Fix this by only enforcing that upstream sends us application/x-rtp
caps as was the case originally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>
Negative composition time offsets are only allowed with version 1 of the
box, however we parse it as a signed value also for version 0 boxes as
unfortunately there are such files out there and it's unlikely to have
(valid) huge composition offsets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
Sometimes the resampler has enough space to store all the incoming
samples without outputting anything. When this happens,
gst_audio_resampler_get_out_frames() returns 0.
In that case, the resampler should consume samples and just return.
Otherwise, we get a segfault when gst_audio_resampler_resample() tries
to resample into a NULL 'out' pointer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1343>
g_sequence_remove_range's end iter is exclusive, so if one
wants to remove that item as well, it should be called with
the next iter.
This could in theory fix an issue where:
* The sequence isn't entirely trimmed, with an old item lingering
* Following FEC packets are immediately discarded because they
arrived later than corresponding media packets, long enough for
seqnums to wrap around
* We now try to reconstruct a media packet with a completely obsolete
FEC packet, chaos ensues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1341>
We need to hold onto the last buffer until the next buffer arrives.
Before, if a caps change comes we would remove the currently rendering
buffer. if Qt asks use to render something, we would render the dummy
black texture.
Fixes a period of black output when upstream is e.g. changing resolution
as in hls adaptive bitrate scenarios.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1338>
The `gst_v4l2_buffer_pool_dqbuf` function contains this ominous comment:
/* get our GstBuffer with that index from the pool, if the buffer was
* outstanding we have a serious problem.
*/
outbuf = pool->buffers[group->buffer.index];
Unfortunately it is common for buffers in _output_ buffer pools to be
both queued and outstanding at the same time. This can happen if the
upstream element keeps a reference to the buffer, or in an encoder
element itself when it keeps a reference to the input buffer for each
frame.
Since the current code doesn't handle this case properly we can end up
with crashes in other elements such as:
(gst-launch-1.0:32559): CRITICAL **: 17:33:35.740: gst_video_frame_map_id: assertion 'GST_IS_BUFFER (buffer)' failed
and:
(gst-launch-1.0:231): GStreamer-CRITICAL **: 00:16:20.882: write map requested on non-writable buffer
Both these crashes are caused by a race condition related to releasing
the same buffer twice from two different threads. If a buffer is queued
and outstanding this situation is possible:
**Thread 1**
- Calls `gst_buffer_unref` decrementing the reference count to zero.
- The core GstBufferPool object marks the buffer non-outstanding.
- Calls the V4L2 release buffer function.
- If the buffer is _not_ queued:
- Release it back to the free pool (containing non-queued buffers).
**Thread 2**
- Dequeues the queued output buffer.
- Marks the buffer as not queued.
- If the buffer is _not_ outstanding:
- Calls the V4L2 release buffer function.
- Release it back to the free pool (containing non-queued buffers).
If both of these threads run at exactly the same time there is a small
window where the buffer is marked both not outstanding and not queued
but before it has been released. In this case the buffer will be freed
twice causing the above crashes.
Unfortunately the variable recording whether a buffer is outstanding is
part of the core `GstBuffer` object and is managed by `GstBufferPool` so
it's not as straightforward as adding a mutex. Instead we can fix this
by additionally recording the buffer state in `GstV4l2BufferPool`, and
handle "internal" and "external" buffer release separately so we can
detect when a buffer becomes not outstanding.
In the new solution:
- The "external" buffer pool release and the "dqbuf" functions
atomically update the buffer state and determine if a buffer is still
queued or outstanding.
- Subsequent code and a new
`gst_v4l2_buffer_pool_complete_release_buffer` function can proceed to
release (or not) a buffer knowing that it's not racing with another
thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1010>
The pipeline flow for receiving looks like this:
rtpsession ! rtpssrcdemux ! session_fec_decoder ! rtpjitterbuffer ! \
rtpptdemux ! stream_fec_decoder ! ...
There are two places where a fec decoder could be placed.
1. As requested from the 'request-fec-decoder' signal: after rtpptdemux
for each ssrc/pt produced
2. after rtpssrcdemux but before rtpjitterbuffer: added for the
rtpst2022-1-fecenc/dec elements,
However, there was some cross-contamination of the elements involved and
the request-fec-decoder signal was also being used to request the fec
decoder for the session_fec_decoder which would then be cached and
re-used for subsequent fec decoder requests. This would cause the same
element to be attempted to be linked to multiple elements in different
places in the pipeline. This would fail and cause all kinds of havoc
usually resulting in a not-linked error being returned upstream and an
error message being posted by the source.
Fix by not using the request-fec-decoder signal for requesting the
session_fec_decoder and instead solely rely on the added properties for
that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1300>
The average_period should always represent the time between two
events. The specification defines the event time as the time
between audio samples, video frame sync, video line sync, etc.
In case of one timestamp per PDU the timestamp_interval identifies
the amount of events between the timestamp of one PDU and the
timestamp of the next PDU.
As described in IEEE 1722-2016 chapter
"10.4.12 timestamp_interval field" timestamp_interval shall be
nonzero.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>
The failure conditions can be overidden by subclasses, and a boolean
return value is provided to the caller whether adding/removing the child
element has actually worked. The caller can then handle this
accordingly but flooding stderr with this is not very useful.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1320>
Upstream caps might for example be
application/x-rtp,media=audio,encoding-name={OPUS, X-GST-OPUS-DRAFT-SPITTKA-00, multiopus}
and while that is not fixed caps it is enough to match it with a media.
Only caps structures that have the correct structure name and that have
the media and encoding-name field are preserved, but if both are present
then these caps are used as "codec preferences".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
Currently reading extension relies on the fact that everything after the
last"." character is a file extension. Whereas that works fine for most
of the cases, it breaks when the URI contains a query part.
E.g.: `http://url.com/file.mp4?param=value` returns `mp4?param=value`
instead of `mp4`.
In this commit we use URI parser to read the path of the URI (in the example
above, that is `/file.mp4`) and read extension from that path.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1305>
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 mpeg2 base class, so that backends that
support render delays can actually implement it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>
Downstream might need the start code offset when decoding.
Previously this computation would be scattered in multiple sites. This
is error prone, so move it to the base class. Subclasses can access
slice->sc_offset directly without computing the address themselves
knowing that the size will also take the start code into account.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>
The GstV4l2CodecAllocator dispose function clears `self->decoder` but
the finalize function then tries to use it if the allocator has no been
detached yet.
Fix by detaching in the dispose function before we clear
`self->decoder`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1220>
Standard interlace handling:
* If we have interlace-mode=interleaved and the field order, we just
set it when creating the session
* If we have interlace-mode=(interleaved|mixed) and no field order, we
set the field order on the first buffer
The encoder session does not support changing the FieldDetail after it
has started encoding frames, so we cannot support mixed streams
correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>
Delay decoders downstream negotiation just before an output frame
needs to be allocated.
This is required, are least for H.264 and H.265 decoders, since
codec_data might trigger a new sequence before finishing upstream
negotiation, and sink pad caps need to set before setting source pad
caps, particularly to forward HDR fields. The other decoders are
changed too in order to keep the same structure among them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>
Using GstBaseDec hack to access the parent_object of each element in
the element itself is a bit fragile. It would be better to keep its
own parent object as the usual global variable. It would make it
resistant to code changes.
The GstBaseDec macro to access the parent object now it's internal to
base decoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>
We were checking if the start time of the gap event was
GST_CLOCK_TIME_NONE, which is superfluous because that cannot happen,
and then not checking if it was NONE after gst_segment_to_running_time,
which caused a crash if an identity received a gap event fully or
partially outside the current segment.
This patch was done in cooperation with:
Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1269>
.. if a current direction has already been set
When `webrtcbin` has created an offer based on codec_preferences,
it might not have received caps on its sinkpads by the time a
remote description is set, in which case we want to connect the
input stream upon actual reception of the caps instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
With mpeg4videoparse drop=false config-interval=N|-1 we might be
trying to insert a config before we have actually received one,
in which case we'll try to map a NULL buffer which will generate
lots of criticals.
Fixes#855
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1265>
gst_va_fixate_format() will iterate all othercaps' structures to find
the one with less information lost at color conversion. If a structure
with same color format is found, the iteration stops. It's like a
smart truncation. Then, this function also will choose the caps
feature.
Later this structure is used fixate its size and no further truncation
is needed.
Don't intersect at fixate, since it kills possible resizing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1261>
If the pad does not have a current caps, get_pad() returns the query
caps which can be ANY. In such case the caps does not have any structure
resulting in a critical warning when calling gst_caps_get_structure().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1254>
Detected while reading the code, cccombiner must set
self->current_video_buffer to NULL *after* emitting selected-samples
in order for the application to get a useful return when peeking
the next video sample.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
When schedule is true (as is the case by default), we insert padding
when no caption data is present in the schedule queue, and previously
weren't checking whether the caption pad had gone EOS, leading to
infinite scheduling of padding after EOS on the caption pad.
Rectify that by adding a "drain" parameter to dequeue_caption()
In addition, update the captions_and_eos test to push valid cc_data
in: without this cccombiner was attaching padding buffers it had
generated itself, and with that patch would now stop attaching
said padding to the second buffer. By pushing valid, non-padding
cc_data we ensure a caption buffer is indeed attached to the first
and second video buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
Also refactor various internals of the monitor code:
- Don't allow starting twice but just return directly when starting a
second time.
- Don't end up in an inconsistent state if call start() a second time
while the monitor is starting up.
- Remove complicated cookie code: it was not possible to add/remove
filters while the monitor was started anyway so this was only useful
in the very small time-window while starting the monitor or while
getting the devices. Instead disallow adding/removing filters while
the monitor is starting, and when getting devices work on a snapshot
of providers/filters.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/667
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1189>
At gst_va_dmabuf_allocator_setup_buffer_full, static code analysis tool
does not know number of objects in descriptor is always larger than 0 if
export_surface_to_dmabuf succeeds. Thus, the tool will assume buf is
allocated with mem but not released when desc.num_objects equals to 0
and raise a mem leak issue.
For gst_va_dambuf_memories_setup, we should also inform the tool that
n_planes will be larger than 0 by checking the value at very beginning.
Then, the defect similar to above will not be raised during static analysis.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1241>
gst_base_parse_reset() does not reset data_bytecount to 0, so
gst_base_parse_update_bitrates() uses a wrong value to calculate
the average bitrate on subsequent pipeline starts. This leads to an
excessive amount of "tag" events being pushed. These events include
very high "bitrate" values that diminish over time, and are produced
until the average bitrate is back to sane values.
Fixes#840
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1228>
This will only affect individual/tarball module builds, as the
options yield to the parent project which was set to gpl=disabled
by default already. We kept it as auto in the original commit
to accommodate the need to update cerbero as well, which had to
be done separately after the initial commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1217>
Previously gst_structure_has_name was used to get a string to compare with supported mimetypes.
This is incorrect as above function returns a user defined structure name which is
not the structure mimetype value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1206>
Trying to reset before the pads have been deactivated races with the
streaming thread. There was also a buggy buffer clear leaving a dangling
`stored_frame` pointer around. Use `gst_interlace_reset` so this happens
properly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1039>
We don't support D3D11 interop for UWP because some APIs
(specifically MFTEnum2) are desktop application only.
However, the code for symbol loading is commonly used by both UWP and WIN32.
Just link GModule unconditionally which is UWP compatible, and simply don't
try to load any library/symbol dynamically when D3D11 interop is unavailable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1216>
... in favour of dep.get_variable('foo', ..) which in some
cases allows for further cleanups in future since we can
extract variables from pkg-config dependencies as well as
internal dependencies using this mechanism.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
When ass hinting value is set to anything other than NONE,
subtitles cannot use smooth scaling, thus all animations will jitter.
The libass author warns about possibility of breaking some scripts when it is enabled,
so lets do what is recommended and disable it to get the smooth scaling working.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1201>
WebVTT in ISO MP4 is specified in ISO 14496-30,
and needed for DASH support. It's stored in an
mp4 specific format. To handle it compatibly,
the wvtt boxes are converted back into WebVTT text
and pushed as application/x-subtitle-vtt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>
Swap the `need_process` boolean check on qtdemux streams
for a direct function pointer to the splitting function,
so we can stop adding extra cases to the single growing
`gst_qtdemux_process_buffer()` function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>
Currently the extension data length specified in the RTP header would
say it was shorter then the data serialised to a packet. When
combining the resulting buffer, the underlying memory would still
contain the extra (now 0-filled) padding data.
This would mean that parsing the resulting RTP packet would potentially
start with a number of 0-filled bytes which many RTP formats are not
expecting.
Such usage is found by e.g. RTP header extension when allocating the
maximum buffer (which may be larger than the written size) and shrinking
to the required size the data once all the rtp header extension data has
been written.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1146>
The code in the aes elements assumes OpenSSL >= 1.1.0:
- implicit library initialization;
- version retrieved with OpenSSL_version(OPENSSL_VERSION);
and it fails to build with older versions.
Specify the required OpenSSL version explicitly in meson.build so that
the elements are excluded on older systems (e.g. Ubuntu 16.04) and the
rest of GStreamer can still build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1067>
An inactive pad is a pad which, in live mode, hasn't yet received
a first buffer, but has been waited on at least once.
Exposing API to support this behaviour allows users of aggregator
subclasses to request pads, and not start pushing data on those
immediately, while avoiding systematic timeouts.
Subclasses must check in explicitly to this behavior, most likely
by exposing a user-facing property, and must check whether a pad
needs ignoring when aggregating. That is because by design,
aggregator subclasses don't get a list of "ready" pads, but instead
directly iterate element->sinkpads.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/867>
Since commit a55dafe341, stream-scoped tags no
longer appeared as top-level tags, introducing a behaviour regression, specially
for MP3 files.
The `gst_discoverer_info_get_tags()` API now returns all tags detected for the
given media, as documented.
A new API is introduced to get container-specific tags,
`gst_discoverer_container_info_get_tags()`. The discoverer tool was adapted to
use it. `gst_discoverer_info_get_tags()` is now deprecated in favor of
`gst_discoverer_container_info_get_tags()` and
`gst_discoverer_stream_info_get_tags()`.
Fixes#759
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1107>
gst_va_vpp_complete_caps_features() now receives the @feature_name to
add and return if @caps doesn't provide it.
So, instead of two nested loops, now the function is a single loop,
traversing @caps to find if each structure already contains the requested
@features_name.
It's important to add missing caps features with @caps, in order to
not lost information.
The function caller does the external loop by calling per each
available caps feature.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1024>
In order to make more readable the caps transformation, the operation
was split in two phases:
1. Rangify the supported caps structures.
2. Add the missing (and supported) caps features.
Step 1 modified its logic, by copying any unrecognized structure.
It's a previous step required for allowing ANY caps feature as
passthrough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1024>
For monorepo build and ugly/bad, for advanced feature
option API like get_option('xyz').required(..) which
we use in combination with the 'gpl' option.
For rest of modules for consistency (people will likely
use newer features based on the top-level requirement).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
There are streams in the wild that have to add a SCTE-35 trigger in
another e.g. GA94 stream. Most encoders would replace the GA94
descriptor ID with the CUEI one temporarily, but there are some that
will add two registration ID descriptors, one with GA94 and one with
CUEI.
Failing to parse the CUEI registration ID in that case would return
FALSE in _stream_is_private_section , therefore setting it as known PES
and pushing packets downstream instead of calling handle_psi.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/979>
In the encoded streams we might not have all the information about the
raw video stream, but when reencoding they end up being specified, even
if those are default values.
As vp8 decoders always output frames in some YUV color space we can
ensure that when upstream doesn't specify any value in its caps we
use the default one which is what we end up doing when decoding/reencoding
anyway, so this way downstream (matroskamux in that case) doesn't need
to be able to renegotiate (which it doesn't).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1062>
We should also take into account whether data is currently pending when checking
for gap on streams. It could very well be that some streams have very low
bitrate (and spread out) data. For those we don't want to push out a gap event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1179>
This is only enabled in push time mode. Furthermore it's only enabled for now if
PCR is to be ignored.
The problem is dealing with streams where the initial PTS/DTS observation might
be greater than following ones (from other PID for example). Before this patch,
this would result in sending buffers without any timestamp which would cause a
wide variety of issues.
Instead, pad segment and buffer timestamps with an extra
value (packetizer->extra_shift, default to 2s), to ensure that we can get valid
timestamps on outgoing buffers (even if that means they are before the segment
start).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1179>
Introduces a `libraries` variable that contains all libraries in a
list with the following format:
``` meson
libraries = [
[pkg_name, {
'lib': library_object
'gir': [ {full gir definition in a dict } ]
],
....
]
```
It therefore refactors the way we build the gir so that we can reuse the
same information to build them against 'gstreamer-full' in gst-build
when linking statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
Since 547570cd79 we do not always build
PyGObject and our development environment is broken when trying to use
GStreamer python when built against system PyGObject with the following
error importing Gst in there:
```
12345678** (gst-plugin-scanner:710617): CRITICAL **: 11:45:02.343: can't find gi.repository.Gst
Traceback (most recent call last):
File "/usr/lib/python3.9/site-packages/gi/repository/__init__.py", line 23, in <module>
from ..importer import DynamicImporter
File "/usr/lib64/python3.9/site-packages/gi/importer.py", line 33, in <module>
from .overrides import load_overrides
ImportError: cannot import name 'load_overrides' from 'gi.overrides' (/var/home/thiblahute/devel/gstreamer/gstreamer/subprojects/gst-editing-services/bindings/python/gi/overrides/__init__.py)
Factory Details:
```
The approach to fixing it is to implement override `gi` in
`gst-python/gi/` which we add to `PYTHONPATH`) and in there reset the
`gi` module to the right place and we get overrides from paths from
`_GI_OVERRIDES_PATH` we set in `gst-env.py` which points to all the
overrides that will be installed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1155>
Making a null check in gst_va_decode_picture_free () indicates pic->buffers or pic->slices
can be null, then in _destroy_buffers () the pointers are dereferenced, which is detected
as dereference after null check by Coverity. Thus, modify the code to do null check in
_detroy_buffers ().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1143>
Retrieving the pad template caps from a ghost pad returns ANY which when
merged with any other caps will return ANY. ANY is not very specific
and may cause suboptimal code paths in e.g. decoders that assume the
lowest common denominator when presented with ANY caps.
Fixes negotiating dma-buf with vaapidecodebin between glupload in the
video sink element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1144>
And use a simple GStreamer pipeline as testsrcbin with GstTranscoder
doesn't let us easily set the framerate of the source and we end up
having videorate dropping frames leading to the rendered file having
an unprecise duration.
This should fix races with `check.gst-editing-services.pythontests.pyunittest.python.test_assets.TestTimeline.test_reload_asset`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1130>
The first approach to fixate was simply a copy&paste of both
videoconvert and videoscale, trying to keep their logic as isolated
as possible. But that brought duplicated and sparse logic.
This patch merge both approaches simplifying the fixate operation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1109>
Add a helper function to get, from GstVideoInfo and GstBuffers flags,
the VA interlace surface flags. This is used currently by vainterlace
element, but it will be used in vapostproc too if it can process
interlaced frames.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1109>
Guard against the orientation not coming from an inexistant tag, nor
from the application (rotation set to "auto") which caused an assertion.
When the application requests the auto rotation method, make sure it is
resolved to a rotation that's applicable.
ERROR:gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstglwidget.c:745:gtk_gst_gl_widget_set_rotate_method: code should not be reached
Fixes: 103ceb853a
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1134>
The src and sink elements no longer link against libsoup. It is now loaded at
runtime. If any version is resident already, it is used. Otherwise we first try
to load libsoup3 and if it's not found we fallback to libsoup2.
For the unit-tests, we now build one version of the test unit file per libsoup
version found. So if both libsoup2 and libsoup3 are available on the host, the
CI will cover them both.
Based on initial patch by Daniel Kolesa <dkolesa@igalia.com> and
Patrick Griffis <pgriffis@igalia.com>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1044>
Actually extract the .o objects from the convience libraries and put
them into the main one. Without this, they will just be referenced by
the .pc file, but it will be unusable because they are not installed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1122>
This new class is a helper for fast/tricky copy of surfaces. First it
tries to copy using the function vaCopy in libva 1.12. If it fails, or
it's not available, a GstVaFilter is tried to be instantiated with the
allocator's parameters, and if succeed, it's used for copying the
source surface.
This is required for dmabuf surfaces with drm modifier.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1023>
Initially GstVaSample processed its GstBuffer member to get the
VASurfaceID. But it might cases where we already have the VASurfaceID
to process by the filter.
This patch enables the possibility to pass the surfaces rather than
the buffers. In order to validate the surfaces a function to check
surfaces were added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1023>
If the USE_PLAYBIN3=1 env var is set, we want to replace
playbin with playbin3, but separate to that, we always
want to register the 'playbin3' element so that applications
which explicitly use playbin3 work regardless of the env var.
This fixes `USE_PLAYBIN3=1 gst-validate-launcher`, for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1102>
... instead of index of DXGI adapter.
The order of IDXGIAdapter1 enumerated via IDXGIFactory1::EnumAdapters1
can be varying even there's no rebooting in case that GPU preference order
is updated by user (for example, it can be done by using NVIDIA Control Panel
in case of multi-GPU laptop system) and eGPU is another possible case.
So, for an element which requires fixed target GPU requirement,
index based device enumeration is unreliable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1098>
* gst_d3d11_device_new_for_adapter_luid()
Used for creating D3D11 device for a DXGI adapter (i.e., GPU)
corresponding to a LUID (Locally Unique Identifier).
This method can be useful for interop with other APIs such as
Direct3D12, MediaFoundation, CUDA, etc.
* gst_d3d11_device_new_wrapped()
Allows creating a new GstD3D11Device object by using already
configured ID3D11Device. This is conceptually equivalent to
gst_gl_context_new_wrapped()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1098>
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>
The old binary/url is no longer available, and the layout of the
sourceforge archvie seems to have changed to include a new
old_versions/ directory where the previous binary is
relocated.
https://sourceforge.net/projects/winflexbison/files/old_versions/
While we could use that, it seem better to invest the effort to
adapt the scipt afterwards to use the github archives instead
https://github.com/lexxmark/winflexbison/releases
So for now bump the version to get the subproject building
till we switch to github releases.
This to simplify the checks made when for example a flex binary
is available system wide: the build process was failing when
that system wide binary didn't have the required version. Instead
of adding more checks and making things more complex, let's just
always use our binary subproject.
libpsl is a dependency if libsoup that has an upstream Meson build
system (in master only).
libsoup also needs libxml2 but we already have a wrap file for it.
The only remaining dependency that must come from the system is sqlite3.
Meson was literally trying to check out tag/revision
'a1b3f07c5271f312997fcc3451237031444c4475 # 1.8.0 + fix for gcc 8.'
which doesn't exist of course.
The proxy-libintl meson build files have been upstreamed, so we do not
need to use Centricular's git repository anymore.
Glib has moved to GNOME's Gitlab instance, and we use a specific branch
on it to get override_find_program for glib tools.
