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>
