The --atleast-version implies --exists, but the implementation in
earlier commits had the version check applied any time the --exists was
checked, and the default value of the major and minor versions were set
to the GStreamer major and minor versions. The resulting behavior would
have gst-inspect return '1' if the plugin's version didn't match
gstreamer's even when --atleast-version was not specified in the command
line args. The change in this patch removes that behavior and adds
tests to verify that if --exists is specified WITHOUT --atleast-version
the version check will NOT be applied. If both arguments are specified
and the version does not match the arg-supplied version number, a new
return code of '2' is used to uniquely identify the failure.
Fixes#3246
Signed-off-by: Thomas Goodwin <thomas.goodwin@laerdal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6191>
This can be used to store informational messages, errors or
warnings which can later be shown to the user in gst-inspect-1.0,
which can be useful for plugins that expose elements dynamically
based on external libraries or hardware capabilities.
Status messages can then provide an indication as to why a
plugin doesn't have any elements listed, for example.
Plus unit test to make sure code paths are exercised a little.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3832>
This allows metas to be serialized to be transmitted or stored. This is
intended to be used for example by gdppay or unixfdsink.
Implemented on GstCustomMeta, GstVideoMeta, GstReferenceTimestampMeta,
and GstAudioMeta.
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5355>
Test included.
The problem appears when aggregator drops the query while
it's being proccessed by the klass->sink_query handler.
This can happen on FLUSH_START event. If the query is still
in the queue, it can be safely dropped, but if it's already
in the klass->sink_query() handler, then sink pad has no
choice and has to wait for the proccessing to complete.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5765>
This is to fix an infinitely blocked upstream streaming thread if
* upstream has fixed-size buffer pool, some H/W decoders for example
* downstream returned flow error without releasing buffer
When the fixed-size buffer pool hits its configured max-buffers and
also downstream of queue returned flow error without releasing corresponding
buffer, upstream has no chance to run the next processing loop
because it will be blocked by acquire_buffer(), and therefore
downstream flow will not be propagated to upstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5023>
Adds gst_queue_array_sort for sorting and gst_queue_array_push_sorted{,struct} for pushing in a sorted order.
All three functions accept a comparison GCompareDataFunc along with optional user_data to pass to it.
In gst_queue_array_sort a small workaround was needed to correctly sort non-struct arrays. Like what _find() already
does, we need to dereference our pointers first, to make sure we can use the same comparison functions everywhere.
This is done via a small wrapper around the provided comparison function.
The array can also wrap around (tail ends up 'before' the head), in which case we have to reorder the array (similar to
what do_expand() does) to then be able to use an existing sorting function, like g_qsort_with_data().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5112>
Fixes test: validate.uridecodebin.expose_raw_pad_caps
testsrcbin (currently part of debugutilsbad) is an useful element for
validate tests.
validate.uridecodebin.expose_raw_pad_caps makes use of it.
Unfortunately, because validate tests with GStreamer only run with
whitelisted plugins and `debugutilsbad` wasn't in the whitelist, the
test was failing and being auto-skipped.
This patch adds debugutilsbad to the whitelists used by validate tests
in subprojects with a validate/meson.build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4931>
This adds code to detect when the hex form of the string we are to
parse exceeds the number of bytes that would form a 32bit flag. This will
avoid treating as flagset anything above then the expected 32 bits and also
stop treading DRM format with modifiers as flagset (like
drm-format=AB24:0x0100000000000002).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4775>
Allow a project to use gstreamer-full as a static library
and link to create a binary without dependencies.
Introduce the option 'gst-full-target-type' to
select the build type, dynamic(default) or static.
In gstreamer-full/static build configuration gstreamer (gst.c)
needs the symbol gst_init_static_plugins which is defined
in gstreamer-full.
All the tests and examples are linking with gstreamer but the
symbol gst_init_static_plugins is only defined in the gstreamer-full
library. gstreamer-full can not be built first as it needs to know what plugins
will be built.
One option would be to build all the examples and tests after
gstreamer-full as the tools.
Disable tools build in subprojects too as it will be built at the end of
build process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
check_version(1.23.1) would return TRUE for a git development version
like 1.23.0.1, which is quite confusing and somewhat unexpected.
We fixed this up in the version check macros already in !2501, so this
updates the run-time check accordingly as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4513>
When copying a buffer, for example with gst_buffer_make_writable(), the
new buffer might reference the same GstMemory as the src buffer,
making those memories not writable. If the src buffer gets disposed
first it should return to its buffer pool, but since some of its
memories are not writable it gets discarded and new buffer/memory gets
allocated.
Solves this by making the new buffer keep a reference to the src buffer,
that ensures that by the time the src buffer gets disposed no other
buffer are referencing its memories and it can thus return safely to its
pool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4176>
gst_buffer_add_parent_buffer_meta() is used when a GstBuffer uses
GstMemory from another buffer that was allocated from a pool. In that
case we want to make sure the buffer returns to the pool when the memory
is writable again, otherwise a copy of the memory is created. That means
the child buffer must drop its ref to the memory first, then drop the
ref to parent buffer so it can return to the pool when it is the only
owner of the memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4176>
This fixes simplification of caps with GstFractionRange structures,
for example, this caps:
video/x-raw, framerate=(fraction)5/1; video/x-raw, framerate=(fraction)[ 5/1, 30/1 ]
can now be simplified to:
video/x-raw, framerate=(fraction)[ 5/1, 30/1 ]
instead of:
video/x-raw, framerate=(fraction){ 5/1, [ 5/1, 30/1 ] }
And this:
video/x-raw, framerate=(fraction)[ 2/1, 5/1 ]; video/x-raw, framerate=(fraction)[ 5/1, 30/1 ]
can be simplified to:
video/x-raw, framerate=(fraction)[ 2/1, 30/1 ]
instead of
video/x-raw, framerate=(fraction){ [ 2/1, 5/1 ], [ 5/1, 30/1 ] }
This fixes overly-complicated GL caps set by avfvideosrc on macOS and
iOS when capturing from a webcam.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4132>
A flush is resetting or not depending on the reset_time argument in the
FLUSH_STOP event is set.
Resetting flushes reset the running time to zero and clear any existing
segment. These are the kind of flushes used by flushing seeks, and by far the
most common. Non-resetting flushes are much more niche, used for instance for
quality changes in adaptivedemux2 and MediaSource Extensions in WebKit.
A key difference between the seek use case and the quality change use case is
that the latter is much more removed from the player. Seeks generally occur
because an user request it, whereas quality changes can be automatic.
Currently, there are three notable cases where position queries fail:
(a) before pre-roll, as there is no segment yet. This is one is understandable,
as for at least some time before pre-roll, we cannot know if a media stream
would start at 0 or any other position, or the duration of the stream for that
matter.
(b) after a resetting flush caused by a seek. This kind of flush resets the
segment, so it's not surprising position queries fail. This is inconvenient for
applications, as it means they always need to handle position reporting (e.g.
in UI) separately every time they request a seek, e.g. by caching the seek
target and using it when the position query fail. I'm not fond of this
behavior, as it's unintuitive and makes GStreamer harder to use, but at this
point could be difficult to change and it's not within the scope of this
proposal.
(c) after a non-resetting flush, e.g. caused by a quality change. The segment
is not reset in this case. Position queries work until a FLUSH_STOP is sent.
Querying position after a FLUSH_START but before a FLUSH_STOP works, and
returns the position the sink was at the moment the FLUSH_START was received.
**This in fact the only reliable way (short of adding probes to the sink
element) to get this position**, as FLUSH_START receival is asynchronous with
playback.
In the case (c), as of currently, position queries fail once the FLUSH_STOP is
received. But unlike in (b), the application has no position to fall back to,
as the FLUSH_START was initiated by elements inside the pipeline that are in a
lower layer of abstraction. Specific applications that have control of both the
player and the internal element doing the flushing -- such as WebKit -- can
still work around this problem through layer violations (lucky!), but this
still puts in question this behavior in GStreamer.
This patch fixes this case by amending the position query handler of basesink,
which was previously erroneously returning early with "wrong state", even
though the flush occurs in PAUSED or PLAYING.
A unit test checking this behavior has also been added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3471>
gst_element_add_pad() is supposed to activate the pad if the element
state is >= PAUSED and the pad is not already active.
Unfortunately, before this patch, the activation was performed while the
element lock was still taken, which ended causing a deadlock in
gst_pad_start_task() as it attempted to post `stream-status` message in
the element, which also requires the element lock.
Elements could work around this bug by activating the pad manually
before adding it to the element.
This patch fixes the problem by performing pad activation only after the
element lock has been released.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3635>
The event type for instant-rate-change events was poorly chosen,
leading to them being re-sent too late and even after EOS.
Add a mechanism in GstPad for the sticky event order to be
different to the value of the event type to fix that up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3387>
Setting force_live lets aggregator behave as if it had at least one of
its sinks connected to a live source, which should let us get rid of the
fake live test source hack that is probably present in dozens of
applications by now.
+ Expose API for subclasses to set and get force_live
+ Expose force-live properties in GstVideoAggregator and GstAudioAggregator
+ Adds a simple test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3008>
Use gst_debug_set_threshold_from_string's new reset behavior to undo
GST_DEBUG and ensure the logging tests have a known configuration.
`gst_debug_set_threshold_from_string ("LOG", TRUE)` has the same effect
as `gst_debug_set_threshold_from_string ("", TRUE)` followed by
`gst_debug_set_default_threshold (GST_LEVEL_LOG)`.
Don't bother remembering the default log level set when the test
started. It will get reset by the next test, anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/605>
These tests relied on setting the name of an element twice to verify
that the last one set took precedence, however name is a CONSTRUCT property
and the parser now errors out when such properties are set twice, in
g_object_new_with_properties .
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3026>
The parent refcount is of the *transformed* buffer, not the input
buffer.
Also update the docs to clarify that @transbuf is the transformed
buffer, and not the buffer on which a transformation is being
performed.
Due to this bug, modifying the structure of a meta that has been
copied to another buffer fails with:
gst_structure_set: assertion 'IS_MUTABLE (structure) || field == NULL' failed
Add a test for the same.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2890>
The latency messages are non-deterministic and can arrive before/after
async-done or during state-changes as they are posted by e.g. sinks from
their streaming thread but bins are finishing asynchronous state changes
from a secondary helper thread.
To solve this, expect latency messages at any time and assert that we
receive one at some point during the test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2643>
Previously it was only possible to request them with the exact template
name, e.g. 'src_%s', but not with "instantiated" names that would match
this template, e.g.'src_foo_bar'.
This is now possible and a test was added for this, in addition to
fixing a previously invalid test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2635>
