For artificial input (in unit tests), all six bytes of
constraint_indicator_flags in hevc_caps_get_mime_codec() can be
zero. Add a guard against an out-of-bounds error that occurred in that
case. Change variables to signed int so comparison with -1 works.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1677>
Duplicating a picture what was already a dup was leading to a crash. Rename
the custom picture flags as HOLDS_BUFFER to make its meaning clear. Then save
then ref and store the picture as userdata, so it can be obtained when
duplicating. Finally, mark the doplicated as HOLDS_BUFFER to avoid thinking it
holds a request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1681>
This patch adds a new parameter: hdr-tone-mapping (same as
vaapipostproc), if the HDR capabilites are availabe in driver, and
it's disabled by default.
If hdr-tone-mapping is enabled then HDR fields in sink caps are
processed in frames from HDR to SDR, removing those hdr fields in
source pad caps too.
hdr-tone-mapping is not enabled if a color conversion is also
requested, since it fails to process in the iHD driver, so far.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1258>
1. Use api_version variable rather than static string.
2. Remove pkgconfig generation since currently the library
is not installed, only used internally.
3. Rely on dependency "required" to abort compilation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1650>
In commit e699aaeb we moved linking of libgudev to the plugin rather
the library, because it's only used in the plugin. But the dependency
check is still done in library.
This patch removes the dependency check in library, and updates the
dependency check in plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1650>
Constantly updating the ts_offset results in audiable glitches
when streaming audio using ntp-sync=true. By requiring a minimum
offset before updating ts_offset this can be mitigated. Added a
parameter which can be used to set min_ts_offset in ntp-sync mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
A new implementation of Intel Quick Sync Video plugin.
This plugin supports both Windows and Linux but optimization for
VA/DMABuf is not implemented yet.
This new plugin has some notable differences compared with existing
MSDK plugin.
* Encoder will expose formats which can be natively supported
without internal conversion. This will make encoder
control/negotiation flow much simpler and cleaner than
that of MSDK plugin.
* This plugin includes QSV specific library loading helper,
called dispatcher, with QSV SDK headers as a part of this plugin.
So, there will be no more SDK version dependent #ifdef in the code
and also there will be no more build-time MSDK/oneVPL SDK
dependency.
* Memory allocator interop between GStreamer and QSV is re-designed
and decoupled. Instead of implementing QSV specific allocator/bufferpool,
this plugin will make use of generic GStreamer memory
allocator/bufferpool (e.g., GstD3D11Allocator and GstD3D11BufferPool).
Specifically, GstQsvAllocator object will help interop between
GstMemory and mfxFrameAllocator memory abstraction layers.
Note that because of the design decision, VA/DMABuf support is not made
as a part of this initial commit. We can add the optimization for Linux
later once GstVA library exposes allocator/bufferpool implementation as
an API like GstD3D11.
* Initial encoder implementation supports interop with GstD3D11
infrastructure, including zero-copy encoding with upstream D3D11 element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1408>
There's no need to do this, and it can make seeking far less accurate.
For a specific use case: I am working with a long (45-minute) MPEG-1 layer 3 file, which has a constant bit rate but no seeking tables. Trying to seek the pipeline immediately after pausing it, without the ACCURATE flag, to a location 41 minutes in, yields a location that is potentially over ten seconds ahead of where it should be. This patch improves that drastically.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/374>
During dispose the pool will still have a reference count of 1 and all
API on it can still be safely called.
Subclasses will have already freed their own data before finalize is
called but would nonetheless be called into again via the pool
deactivation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1645>
It's almost pointless and makes little sense as subclass might
want to modify refcount of the object or so. And all subclasses
are already casting them to non-const version as well.
In a general sense, we need to avoid passing refcounted object
with const qualifier.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1238>
... in order to make older g-i happy (~1.60) which doesn't like
freeform descriptions in the value_name field. Which in turn
then makes hotdoc happy instead of erroring out when we bump
the symbol index version.
We usually only (ab)use the name field for description strings
for private plugin enums, not for public API visible to bindings.
This lets glib-mkenum generate the _get_type() function for the
enum again, which in turn will generate the expected value names
to match the enums.
We might be able to add this back later once we can upgrade the
g-i version requirement (and the documentation job image).
This reverts most of commit b0aab48cdcf0a454d14aeb4d907209d8ee3f1add
There's a race condition in gsttagdemux.c between typefinding and the
end-of-stream event. If TYPE_FIND_MAX_SIZE is exceeded,
demux->priv->collect is set to NULL and an error is returned. However,
the end-of-stream event causes one last attempt at typefinding to occur.
This leads to gst_tag_demux_trim_buffer() being called with the NULL
demux->priv->collect buffer which it attempts to dereference, resulting
in a segfault.
The malicious MP3 can be created by:
printf "\x49\x44\x33\x04\x00\x00\x00\x00\x00\x00%s", \
"$(dd if=/dev/urandom bs=1K count=200)" > malicious.mp3
This creates a valid ID3 header which gets us as far as typefinding. The
crash can then be reproduced with the following pipeline:
gst-launch-1.0 -e filesrc location=malicious.mp3 ! queue ! decodebin ! audioconvert ! vorbisenc ! oggmux ! filesink location=malicious.ogg
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/967
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1620>
The kCVPixelFormatType_64RGBALE enum is only available on macOS Big
Sur (11.3) and newer. We also cannot use that while configuring the
encoder or decoder on older macOS.
Define the symbol unconditionally, but only use it when we're running
on Big Sur with __builtin_available().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1613>
Hotdoc should be able to extract and parse comments out of these. Just
need to be careful to only add the glob in directories that actually
contain *.m (objc) and *.mm (objcpp) files.
Also fix some doc comments and remove redundant ones.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1614>
Platform wise, is not possible, as far as I known, to have Wayland
without kernel's DRM. Though, it's possible to configure
gstreamer-vaapi without DRM but Wayland support, with the enhanced
handling of dmabuf in vaapisink for Wayland, vaapisink will always
fail. Given both issues, configuration with no DRM but Wayland, makes
things more complex, and a simpler approach is to refuse that
configuration.
This patch disables Wayland support if there isn't DRM support. Also,
it disables the display test for Wayland, relying only on DRM and
X11.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1606>
Formats map is instantiated at the end of the display
instantiation. The problem is the Wayland display which looks for a
format in a callback, before the map is populated.
If user compiles gstreamer-vaapi with DRM support, the map is
populated with a DRM display at GStreamer plugin registration. But if
not, or a VA driver is not available, the plugin will try with a
Wayland driver, which cause the NULL de-reference.
Nevertheless, in the case of no DRM support, and if the Wayland
display doesn't get a reply from the format conversion is not a
problem.
So the solution is the trivial one, check if the format map is already
populated before de-reference it.
Fixes: #977
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1606>
This is listed as a public interface implemented by osxaudio, so we
need to mark it as a plugin API so that it's listed in the
documentation correctly.
This is an ancient symbol, so add it to the symbol index too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1601>
These symbols from macOS plugins osxaudio, osxvideo, and applemedia
have been present for a very long time but were never documented.
This allows us to document these, and also add Since: markers for new
features (symbols) there were added in 1.20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1601>
This moves the ABI check to the registration, so we don't expose
decoders with the wrong ABI or that are just broken somehow. It
also makes few enhancement:
- Handle missing, but required controls
- Prints the controls macro name instead of id
This should fix RK3399 support with a currently release minor
regression in the Hantro driver that cause errors.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1599>
... in addition to all the other issues that were ignored for them
already.
At least the reverse playback test regularly times out, waiting for a
download to finish that has already finished successfully.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1593>
Starting with libsoup3, there is no attempt to handle thread safety
inside the library, and it was never considered fully safe before
either. Therefore, move all session handling into its own thread.
The libsoup thread has its own context and main loop. When some
request is made or a response needs to be read, an idle source
is created to issue that; the gstreamer thread issuing that waits
for that to be complete. There is a per-src condition variable to
deal with that.
Since the thread/loop needs to be longer-lived than the soup
session itself, a wrapper object is provided to contain them. The
soup session only has a single reference, owned by the wrapper
object.
It is no longer possible to force an external session, since this
does not seem to be used anywhere within gstreamer and would be
tricky to implement; this is because one would not have to provide
just a session, but also the complete thread arrangement made in
the same way as the system currently does internally, in order to
be safe.
Messages are still built gstreamer-side. It is safe to do so until
the message is sent on the session. Headers are also processed on
the gstreamer side, which should likewise be safe.
All requests as well as reads on the libsoup thread are issued
asynchronously. That allows libsoup to schedule things with as
little blocking as possible, and means that concurrent access
to the session is possible, when sharing the session.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/947
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1555>
Intel drivers expose some colorbalance's maximum values much more
bigger than their minimum values, given their middle values (default
value). This means, in practice, that the real middle point between
the maximum and minimum values implies a major change in the color
balance, which is not expected by the GStreamer color balance logic.
This patch makes the given maximum value symmetrical to the minimum
value, given the middle one (default value).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1580>
It was getting pulled in automatically via cairo, but the version
there is too old for us. We need the latest to fix Windows ARM64 NEON
support:
> ERROR: No specified compiler can handle file subprojects\libpng-1.6.37\arm/filter_neon.S
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1570>
As now, we warn if the decoder have no support src pixel format, but that
warning is called before the type (hence the debug category) is initialized.
Fix this by moving the debug category init out of the type initialization,
into the register funcitons.
This will fix an assertion that occures in the register function and allow
relevant log to be seen by the users.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1588>
Those will cause us to renegotiate at the next aggregate cycle,
and while at that point we may decide to reconfigure upstream
branches (in practice we don't as this is inherently racy,
and that's the reason why mixer subclasses perform conversion
internally), we certainly don't want to just forward the event
willy-nilly to all our sinkpads.
An actual issue this is fixing is when caps downstream of a
compositor are changed at every samples-selected signal emission,
for the purpose of interpolating the output geometry, and the
compositor has a non-zero latency, the reconfigure events were
forwarded to basesrc, which triggered an allocation query, which
in turn caused aggregator to have to drain (thus not being able
to queue <latency> frames), leading to disastrous effects
(choppy output as compositor couldn't consume frames fast enough,
the higher the latency the choppier the output)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1464>
The merge request workflow can be confusing to people unfamiliar with
it, so add screenshots.
Also add a new section on how to revise merge requests, since a lot of
people tend to open new merge requests to make any changes.
Eliminate the separate "How to Prepare a Merge Request for Submission"
section -- merge it with the main section.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1367>
This is usually necessary to allow gst-indent to treat it as
a statement, but we do not run gst-indent on headers and we do not
have extra semicolons in other places that this macro is used in the
header. Fixes warnings when using the header:
```
In file included from gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/video.h:185,
from XYZ:9001:
gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h:206:78: warning: ISO C does not allow extra ‘;’ outside of a function [-Wpedantic]
206 | G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstVideoAggregatorConvertPad, gst_object_unref);
| ^
gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h:214:181: warning: ISO C does not allow extra ‘;’ outside of a function [-Wpedantic]
214 | G_DECLARE_DERIVABLE_TYPE (GstVideoAggregatorParallelConvertPad, gst_video_aggregator_parallel_convert_pad, GST, VIDEO_AGGREGATOR_PARALLEL_CONVERT_PAD, GstVideoAggregatorConvertPad);
| ^
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1572>
When fixating color, there might be "other caps" with color spaces not
supported by the caps features exposed in the vapostproc's source pad
caps template (perhaps it's a bug somewhere else in GStreamer).
This solution checks if the proposed format exists in the filter
within the caps feature associated with the proposed format.
The check is done with the new filter's function
gst_va_filter_has_video_format().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1559>
In the need-data appsrc callback, a buffer is pulled from the
appsink. This buffer is then copied so that metadata is writable.
The copy is pushed to the appsrc but it doesn't take ownership
of the buffer so we need to manually unref it. The original buffer
is finally unreffed when the sample is freed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>
This is due to an unsafe usage of the pad task. We didn't ensure proper
ownership of the task. That race involved the task being released too early,
and was detected, luckily, by the glib mutex implementationt that
reported the mutex being disposed while being locked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1478>
On Android (especially) and for static builds in general it is safer to link
against libsoup and have the dynamic custom loading disabled. For those cases we
can safely assume the application will use either libsoup2 or libsoup3 and not
both.
Fixes#939
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1536>
The earlier size of 2 MB was set back in 2009, it doesn't
seem unreasonable to raise it to 8 MB these days. The use
case at hand is matroskademux containing both a video stream
with a very low amount of compression but no decoding latency,
and a H265 stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1538>
It is an extremely common mistake on Windows to have incorrect PATH
values when loading a plugin, and the error from g_module_error()
(which just calls FormatMessageW()) is very confusing in this case:
The specified module could not be found.
https://docs.microsoft.com/en-us/windows/win32/debug/system-error-codes--0-499-#ERROR_MOD_NOT_FOUND
It implies the plugin itself could not be found. The actual issue is
that a DLL dependency could not be found. We need to detect this case
and print a more useful error message.
We should still print the error fetched from FormatMessage() so that
people are able to google for it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1540>
GstAudioRingBufferSpec::segsize has been configured by using
device period but GstWasapi2RingBuffer was referencing the
buffer size returned by IAudioClient::GetBufferSize()
which is most likely larger than device period.
Fixing to sync them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1533>
If the `area_surface` got unmapped when changing to the `READY` or
`NULL` state, we currently don't remap it when playback resumes and
`wp_viewporter` is supported. Without `wp_viewporter` we do remap
it, but rather unintentionally and also when not wanted.
On Weston this has not been a big problem as it so far wrongly maps
subsurfaces of unmapped surfaces anyway - i.e. only the black
background was missing on resume. On other compositors and future
Weston this prevents the `video_surface` to get remapped.
Shuffle things around to ensure `area_surface` is mapped in the
right situations and do some minor cleanup.
See also https://gitlab.freedesktop.org/wayland/weston/-/issues/426
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1483>
Latest ffmpeg has removed avcodec_get_context_defaults(), and its
documentation says a new AVCodecContext should be allocated for this
purpose. The pointer returned by avcodec_find_decoder() is now
const-qualified so we also need to adjust for it. And, AVCOL_RANGE_MPEG
is now rejected with strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1531>
The later, doing damage in surface coordinates instead of buffer
coordinates, has been deprecated. The reason for that is that it
is more prone to bugs, both on the client and the compositor side,
especially when paired with buffer scale, `wp_viewporter` or
buffer transforms.
Unfortunately, on Weston this risks running into
https://gitlab.freedesktop.org/wayland/weston/-/issues/446
(which causes trouble for several other projects as well). However,
that bug only affects cases where we run in sync mode, i.e. only
during resizes. In practise I haven't been able to observe the
issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
Each time we call `wl_surface_damage()` we want to do full surface
damage. Like Mesa, just use `G_MAXINT32` to ensure we always do
full damage, reducing the need to track the right dimensions.
`window->video_rectangle` is now unused, but we keep it around for
now as we may need it again in the future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
From the spec:
> This request is used to describe the regions where the pending
> buffer is different from the current surface contents
We currently also call `wl_surface_damage()` on surfaces without
new or still compositor-hold buffers, e.g. when resizing the window.
In that case we call it on `area_surface_wrapper`, even though it
gets resized via `wp_viewport_set_destination()`, in which case
the compositor is in charge of repainting the area on screen.
Doing so is currently not forbidden by the spec, however it might
be in the future, see
https://gitlab.freedesktop.org/wayland/wayland/-/issues/267
Thus lets stay close to the spec and only call `wl_surface_damage()`
when we just attached a buffer.
Right now this prevents runtime assertions in Mutter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
`gst_wl_window_set_opaque` does not get called on window resizes,
potentially leaving opaque regions too small.
According to the spec opaque regions can be bigger than the surface
size - parts that fall outside of the surface will get ignored.
Thus we can can simply use `G_MAXINT32` and be sure that the whole
surfaces will always be covered.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
If the VANC track does contain packets, but we skip over all packets, just
treat it the same as if there hadn't been any packets at all and send a
GAP event instead of erroring out with "Failed to handle essence element".
We would error out because when we reach the end of the loop without having
found a closed caption packet the flow return variable is still FLOW_ERROR
which is what it has been initialised to.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1518>
Though the profiles[0] is inited as GST_H265_PROFILE_INVALID in the
gst_h265_profile_tier_level_get_profile(), the profile detecting may
change its content later. So the return of profiles[0] may not be an
invalid profile even the len is 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1517>
The previous code was mistakenly trying to compute a cc_type out
of the first byte in the byte triplet, whereas it is to be interpreted
as:
> Bit b7 of the LINE value is the field number (0 for field 2; 1 for field 1).
> Bits b6 and b5 are 0. Bits b4-b0 form a 5-bit unsigned integer which
> represents the offset
The same mistake was made when creating padding packets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1496>
Sometimes we can't output anything because we don't have enough
incoming frames. In that case, the resampler was trying to call
do_quantize() and do_resample() in a loop forever because there would
never be samples to output (so chain->samples would always be NULL).
Fix this by not calling chain->make_func() in a loop -- seems
completely unnecessary since calling it over and over won't change
anything if the make_func() can't output samples.
Also add some checks for the input and / or output being NULL when
doing conversion or quantization. This will happen when we have
nothing to output.
We can't bail early, because we need resampler->samples_avail to be
updated in gst_audio_resampler_resample(), so we must call that and
no-op everything along the way.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1461>
When the image is opaque but the output ProRes format has an alpha
component (4 component, 32 bits per pixel), Apple requires that we
signal that it should be ignored by setting the depth to 24 bits per
pixel. Not doing so causes the encoded files to fail validation.
So we set that in the caps and qtmux sets the depth value in the
container, which will be read by demuxers so that decoders can skip
those bytes entirely. qtdemux does this, but vtdec does not use this
information at present.
The sister change was made in qtmux and qtdemux in:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1061
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1489>
