In the case of a gstreamer-full target type to static,
the GST_STATIC_COMPILATION is necessary on Windows to avoid
a different mangling from the external project using the
gstreamer-full libraries (ie dllimport).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
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>
According to the documentation this should never happen but apparently
does under certain circumstances. As the sockets are set non-blocking,
trying to read from them regardless should not cause any problems.
In all cases that were observed so far, the socket in question actually
has a packet queued up for reading.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4748>
While this doesn't yet use any OS provided times from the actual network
stack, this still gets rid of any IPC jitter between the helper process
and the main process as part of the PTP time calculations and should
improve accuracy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4665>
On Windows and macOS always use the proper monotonic clock, including
for gst_util_get_timestamp(), and initialize its state only once.
Also on macOS use clock_gettime() for the realtime clock, if available
instead of always falling back to GLib.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4658>
ptpd is defaulting to the hybrid mode, and was sending invalid multicast
PTP messages in that configuration until ce96c742a88792a8d92deebaf03927e1b367f4a9.
While this commit was made in 2015 there was no release in the meantime.
Work around this by detecting this case and defaulting to the default
values for the given intervals as given by the PTP standard.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4654>
Setting the policy to NSApplicationActivationPolicyAccessory by default makes
sure that we can activate windows programmatically or by clicking on them.
Without that, windows would disappear if you clicked outside them and there
would be no way to bring them to front again. This change also allows osxvideosink
to receive navigation events correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4573>
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>
The generated gir file marks the size parameter as "out" by default.
This is wrong in the context of a caller allocated buffer with a given size.
Explicitly marking the size parameter as (in) fixes the issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4399>
This works on Linux, Android, Windows, macOS, FreeBSD, NetBSD, OpenBSD,
DragonFlyBSD, Solaris and Illumos.
Newly supported compared to the C version is Windows.
Compared to the C version various error paths are handled more correctly
and a couple of memory leaks are fixed. Otherwise it should work identically.
The minimum required Rust version for compiling this is 1.48, i.e. the
version currently in Debian stable. On Windows, Rust 1.54 is needed at
least.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1259
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3889>
gst_base_src_new_segment() does not send the segment right away, which
may break events ordering if subclass sends other events after
calling it.
Introducing a variant pushing the segment right away to preserve
ordering in such cases.
Will be used by appsrc which has its own internal queue where we need to
preserve events order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4297>
We don't need to obtain the mutex to ensure that `sq` is non-NULL. `sq`
is assigned immediately after the pads are created and not destroyed
until the pads are finalized.
Use the pad direction to determine which internal peer we need.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/888>
When a pipeline is pre-rolling, it waits for all sink elements to report
they have received a buffer before completing the transition to paused.
This async wait is done using a state condition variable. The way this
waits are currently implemented do not protect against spurious conditional
wake ups, which may happen due to external factors in the kernel.
This change implements the wait within a loop that iterates over the protected
variable to reinitiates the wait if the wakeup was spurious. More details in
the [GCond docs](https://docs.gtk.org/glib/struct.Cond.html).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4086>
It's quite confusing to print a function callback signature for
action signals when people need to do a g_signal_by_name() invocation
in order to use this feature. Requires too much background knowledge
about how GObject works under the hood to make sense of that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4299>
Existing codes rely on modified argc value by g_option_context_parse()
but g_option_context_parse_strv() is used in case of Windows.
Count arguments after the option parsing manually.
Fixing command "gst-inspect-1.0.exe -b"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4313>
Fix compiler warnings about not using the return value when
freeing the GString segment with g_string_free(.., FALSE):
ignoring return value of ‘g_string_free_and_steal’ declared with attribute ‘warn_unused_result’
which we get with newer GLib versions. These were all harmless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4273>
When using such a launch line:
fakesrc ! "audio/x-opus, channel-mapping=(int)<0, 1>" ! fakesink
the caps string, with spaces escaped but no quotes gets passed to
gst_caps_from_string(), which then fails to parse the array because it
contains spaces.
When using an explicit capsfilter instead:
fakesrc ! capsfilter caps="audio/x-opus, channel-mapping=(int)<0, 1>" ! fakesink
the caps string, with spaces escaped and quotes gets passed through
gst_value_deserialize, which first calls gst_str_unwrap() on it and only
then gst_caps_from_string() on the result.
This fixes the inconsistency by using a custom version of str_unwrap()
in the parser, which doesn't expect a quoted string.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4181>
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 patch adds documentation to the 'log' tracer and amends the design
document of Tracers to replace a misleading example of the 'log' tracer
with a different example that uses tracer arguments with tracers that do
actually handle said arguments.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4153>
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>
When we run Cheese 41.1 on our imx platform, Cheese preview freeze
at first frame.
During pipeline state changing from NULL to PLAYING, if there are
both elements that state change asynchronously and state change
with no preroll in the bin, the element inside may send ASYNC_DONE
message to it, while the bin's pending state is VOID_PENDING.
In this case, the bin will not post ASYNC_DONE message to parent
bin, which makes parent bin thinks that there are still elements
in it that haven't completed state changing, causing the pipeline
freeze in an intermediate state.
This commit modifies the bin_handle_async_done() function. When the
bin, whose pending state is VOIDING_PENDING, receives the ASYNC_DONE
message, it will also post this message to its parent bin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3490>
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>
The previous implementation was a bit primitive, assuming the subclass
had registered a template name starting with sink_ . Instead make
the effort of parsing the actual template name, and use that to generate
the final pad name.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4032>
Fixes the following valgrind error:
==616== Conditional jump or move depends on uninitialised value(s)
==616== at 0x4900E34: gst_debug_print_object (gstinfo.c:1143)
==616== by 0x49010B6: gst_info_printf_pointer_extension_func (gstinfo.c:1215)
==616== by 0x4959FDB: __gst_printf_pointer_extension_serialize (printf-extension.c:47)
==616== by 0x495A487: printf_postprocess_args (vasnprintf.c:258)
==616== by 0x495A52C: __gst_vasnprintf (vasnprintf.c:290)
==616== by 0x4959F8F: __gst_vasprintf (printf.c:154)
==616== by 0x4901C1F: gst_debug_message_get (gstinfo.c:791)
==616== by 0x4901C75: _gst_debug_log_preamble (gstinfo.c:1431)
==616== by 0x4903208: gst_debug_log_default (gstinfo.c:1575)
==616== by 0x49020BA: gst_debug_log_full_valist (gstinfo.c:624)
==616== by 0x490211D: gst_debug_log_valist (gstinfo.c:656)
==616== by 0x49021AD: gst_debug_log (gstinfo.c:533)
==616== by 0x48DDC11: gst_buffer_copy_into (gstbuffer.c:693)
==616== by 0x48DF5F1: gst_buffer_copy_with_flags (gstbuffer.c:727)
==616== by 0x48DF640: gst_buffer_copy_deep (gstbuffer.c:756)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4034>
When the task already exists, we forgot to free the passed `user_data`.
This wasn't an issue for most C code, which doesn't pass a
`GDestroyNotify`, but bindings such as gstreamer-rs do!
That said, allocating a trampoline in gstreamer-rs just for it to get
thrown away again is awkward. Maybe we need a `gst_pad_resume_task`?
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3920>
Since b76d336549
pads are deactivated when going to READY but in `uridecodebin(3)`, the
sources source pads are activated while in NULL state (when PULL mode is
supported), meaning that we are ending up deactivating those pads in
NULL_TO_READY, breaking the pipeline.
The intent of the commit mentioned above is to ensure that the pads are
deactivated either in PAUSED_TO_READY or READY_TO_READY, so it should
be safe to avoid deactivating in NULL_TO_READY.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3849>
Usually gst-plugin-scanner.exe will be located under libexec/gstreamer-1.0
or even somewhere user specified location via GST_PLUGIN_SCANNER
environment. So, in order for child process to be able to load
GStreamer DLLs, parent process will need to update PATH env
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3886>
We already have functions to generate a stream-id from pads but in the
end those pads are not even used in most cases. This adds functions to
generate a stream-id even before creating the source pads for the
element that is going to use it. For example a demuxer that is properly
implements the GstStream/GstStreamCollection API will not have a Pad but
already needs to generate a stream-id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3160>
It's not possible to annotate a in-parameter for a return value array as
the array length. Both are assumed to have the same direction and the
current annotation causes the size parameter to be considered an out
parameter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3787>
Due to the dynamic nature of multiqueue, when `use-interleave` is used we can't
report a maximum tolerated latency (when queried) since it is calculated
dynamically.
When in such live pipelines, we need to make sure multiqueue can handle the
lowest global latency (provided by this event). Failure to do that would
result in not providing enough buffering for a realtime pipeline.
Fixes#1732
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3772>
Pads are activated automatically when they are added if the element
state is >=PAUSED, so it's not necessary to activate them manually
anymore.
This patch removes manual pad activation from gstaggregator, gstconcat,
gstfunnel, and gstinputselector.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3636>
As the path to the gir file is passed to hotdoc.generate_doc() and
not the build target itself, meson doesn't know about the dependency.
In turn, as the CI doesn't build everything before building the
documentation target, some gir files might not exist, for instance
in the case of gst-rtsp-server, causing the output documentation to
be empty.
The error occurred silently because hotdoc accepts wildcards for
*-sources arguments, thus it won't warn about a missing gir file as
it is legitimate for glob matching to resolve to nothing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3686>
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>
Commit d3a66f9851 introduced a potential deadlock with two parallel release_pad
calls, where one could release the main multiqueue lock (qlock) while still
holding the reconf_lock and then calling other routines which in some conditions
may try to acquire qlock again. The second release_pad could already acquire the
qlock and then start waiting on reconf_lock, which may never be possible because
because the first one isn't releasing it until it can acquire qlock.
Fix it by holding reconf_lock for the whole durationg of qlock, making this
particular deadlock impossible.
Fixes#1642
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3571>
On macOS, a Cocoa event loop is needed in the main thread to ensure
things like opening a GL window work correctly. In the past, this was
patched into glib via Cerbero, but that prevented us from updating it.
This workaround simply runs an NSApplication and then calls the
main function on a secondary thread, allowing GStreamer to correctly
display windows and/or system permission prompts, for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3532>
gst_plugin_load_by_name() assumed a plugin has a filename,
which isn't true for static plugins, leading to criticals.
If a plugin is already loaded, just return the loaded plugin,
which makes it work for static plugins as well as saving a
moment for already-loaded dynamic plugins.
Add locking in gst_plugin_is_loaded(), as a plugin may be
still being loaded in another thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3552>
This adds "id" variants to most debugging functions, and allows providing a
string identifier instead of a GObject.
This allows providing unified and clearer debug logs for all the
non-gobject-based items, and opens the way for more unified logging.
As an extension, copying the object name is avoided as much as possible, by
using it directly instead of going through another copy.
* API : gst_debug_message_get_object_id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3483>
When coloring is in use, those escape codes are going to be created many times
for almost all debug lines.
Don't create plenty of temporary allocations, and instead build the escape code
ourselves statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3498>
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>
When dealing with gapless input (i.e. streams with changing group-id in
GST_EVENT_STREAM_START), we need to take into account the elapsed
running-time (if applicable) in order to properly calculate levels and output
time. Without doing this all incoming data from future groups would be
considered as being "late" and would be consumed immediately.
This does **NOT** modify the actual segment and buffer times, and is only used
internally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
Introduce a new API that can return a GstTypeFind * with helper functions
and data set around buffer data.
While at it, drop factory field from GstTypeFindBufHelper. While it was
useful for logging, it was not passed through function arguments and keeping
it for logging would require an additional API increasing the API surface
and making it harder to use.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3296>
