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/2357>
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/2327>
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/2322>
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/2321>
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/2320>
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/2301>
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/2297>
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/2240>
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/2236>
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/2236>
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/2208>
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/2167>
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/2168>
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/2162>
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/2162>
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/2118>
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/2035>
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/2074>
../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/2068>
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/2053>
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/2013>
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/1995>
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/1995>
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/1991>
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/1979>
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/1940>
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/1923>
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/1949>
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/1948>
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/1919>
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/1919>
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/1920>
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/1901>
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/1895>
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/1890>
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/1870>
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/1833>
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/1833>
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/1832>
```
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/1832>
```
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/1832>
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/1832>
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/1829>
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/1829>
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/1822>
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/1823>
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/1809>
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/1804>
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/1758>
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/1793>
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/1781>
... 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/1755>
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/1754>
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/1736>
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/1716>
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/1716>
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/1716>
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/1716>
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/1712>
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/1712>
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/1706>
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/1702>
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/1702>
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/1695>
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/1661>
... 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>
