In tests in the rust bindings we end up with 2 thread initializing
concurrently, and it should not be a problem, -validate should be MT
safe.
Using a recursive mutex as we might recursively init for some reason
and we are not on the hot path here in any case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4097>
Recursively invoking the NSMainLoop can cause crashes in
applications that don't expect it. Instead of waiting for
permission to be granted, move the wait later - until we
actually need device permissions when starting the capture
session. That moves the wait into the streaming thread
instead of the application thread that's setting the pipeline
state to READY.
Instead of a manual state change implementation to open
and close the device, use the basesrc start/stop methods that
are intended for the purpose.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4096>
These checks were introduced to prevent exposing ARGB64/RGBA64 in the caps
when running on M1 Pro/Max with macOS <13 because of a bug in VideoToolbox.
Unfortunately, the initial buffer size of 15 is too short when running
in a VM - the CPU brand string there looks like "Apple M1 Pro (Virtual)",
which due to its length causes sysctlbyname to return -1, resulting in
broken formats still showing up in the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4082>
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/4038>
The av1decoder class does not implement the ->parse() virtual function,
and we always need to add the av1parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>
The vp9decoder class does not implement the ->parse() virtual function,
and we always need to add the vp9parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>
The vp8decoder class does not implement the ->parse() virtual function,
it can only accepts frame aligned data. If some element such as filesrc
feed it with unaligned data, the behaviour is undecided. So we should
set_needs_format of the decoder to TRUE, then it can fail with a
"not-negotiated" error early, rather than go on and generate unexpected
error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>
The mpeg2decoder class does not implement the ->parse() virtual function,
and we always need to add the mpegvideoparse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>
The h264decoder class does not implement the ->parse() virtual function,
and we always need to add the h264parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>
The h265decoder class does not implement the ->parse() virtual function,
and we always need to add the h265parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>
These parameters are not actually `out` parameters but must
be allocated and zero-initialized by the calling function.
Marking them as `out caller-allocates` will cause memory
corruptions when calling these APIs from e.g., Python code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4061>
The dimension of the overlay texture directly corresponds to the size of the overlay **buffer** which is given by its video meta.
The dimension at which the overlay should be displayed directly correspond to the overlay `render_width`and `render_height`.
This match the behavior of glimagesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4053>
The av1C box is optional so dropping parsing does not break anything
fundamentally, and there seems to be no historical record how version 0
even looks like while the comments and the parsing disagreed with each
other.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4027>
If we have caps then we can only set exactly those caps, if we have no
caps yet then negotiating anything is not very meaningful because the
caps are defined by the application and not downstream.
Avoids, among other things, an unnecessary allocation query and spurious
useless caps being set before the first buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4020>
This crept in several years ago sadly :(
The usage of accurate seeking should be reserved to use-cases where it is
essential that we seek to that position. This should not be the default.
There is a new option `--acurate-seeks/-a` to be able to force that.
Furthermore, if accurate seeks aren't required, a player should be using the
GST_SEEK_FLAG_KEY_UNIT flag to seek to the closest keyframe and provide the most
reactive experience.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4017>
Generating the source element is done when uridecodebin is doing the
READY to PAUSED state change, so it is reasonable to set the new source
element to that state.
This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4016>
When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).
Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:
ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it
This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.
Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.
Co-authored by: Alicia Boya García <ntrrgc@gmail.com>
...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467
[1] https://github.com/rdkcentral/mvt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3990>
This fixes a compile error with recent upstream FFmpeg.
The AV_CODEC_CAP_AUTO_THREADS was deprecated and renamed to
AV_CODEC_CAP_OTHER_THREADS in FFmpeg upstream commit
7d09579190de (lavc 58.132.100).
The AV_CODEC_CAP_AUTO_THREADS was finally removed in FFmpeg upstream
commit 10c9a0874cb3 (lavc 59.63.100).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3964>
Fixes#1358.
Passing ARGB64/RGBA64 to vtenc caused the encoding to fail
when running on M1 Pro/Max variants with macOS 12.x, so let's
remove these formats from caps when such scenario is detected.
This issue appears to have been fixed OS-side in macOS 13.0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3962>
This was causing incorrect output when seeking, especially
when used with a multithreaded source like `videotestsrc n-threads=2`.
It should now correctly wait for frames still being processed by VT
while vtdec is flushing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3937>
We are using std::isspace() with one parameter. That function is defined
in the cctype header.
```
win32ipcutils.cpp(34): error C2672: 'std::isspace': no matching overloaded function found
win32ipcutils.cpp(34): error C2780: 'bool std::isspace(_Elem,const std::locale &)': expects 2 arguments - 1 provided
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3936>
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/3925>
The VA API has not defined the scaling list entries for U/V planes
for the 4:4:4 stream. In fact, we do not meet the 4:4:4 format output
for H264 so far, and scaling list is not used frequently, so we just
print out some warning and ignore these scaling list values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3877>
The gst-devtools project generates gstreamer-validate-1.0.pc, this
must match the dependency in gst-editing-services for detection
to work properly.
Fixes:
Run-time dependency gst-validate-1.0 found: NO (tried pkgconfig and cmake)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3872>
Due to a bug in the VT API, attempting to encode interlaced content
with ProRes results in an error, halting the pipeline instead of
gracefully falling back to software encoding.
Should be removed in the future if Apple ever fixes this issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3878>
All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.
This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3866>
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/3814>
Spec 7.1.3:
If a memory object does not have the VK_MEMORY_PROPERTY_HOST_COHERENT_BIT
property, then vkFlushMappedMemoryRanges must be called in order to guarantee
that writes to the memory object from the host are made available to the host
domain, where they can be further made available to the device domain via a
domain operation. Similarly, vkInvalidateMappedMemoryRanges must be called to
guarantee that writes which are available to the host domain are made visible to
host operations.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3817>
It is really difficult for people to figure out why nvcodec has
0 features. Even the debug log is cryptic. Also make sure the errors
go to the ERROR log level, which is more likely to be enabled by
default.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3813>
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/3810>
Deserialize socket control messages as GstSocketTimestampMessage only
if (level, type) is (SOL_SOCKET, SCM_TIMESTAMPNS).
Without this patch, messages with types SCM_RIGHTS or SCM_CREDENTIALS
could be deserialized as GstSocketTimestampMessage instead of
GUnixFDMessage or GUnixCredentialsMessage from gio.
Fixes#1736
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3777>
Handling mouse navigation events in glvideomixer element, if no
pixel-aspect-ratio info in the caps, an assertion error is produced
inside gst_util_fraction_multiply because default denominator is zero.
Error fixed:
```
(gst-launch-1.0:102654): GStreamer-CRITICAL **: 00:47:51.598: gst_util_fraction_multiply: assertion 'b_d != 0' failed
```
Simple pipeline to reproduce the issue:
```
gst-launch-1.0 -v glvideomixer name=mix ! glimagesinkelement gltestsrc ! mix.sink_0
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3766>
Allow requesting an offer from the peer if we're joining a call with a
peer, and allow the peer to request an offer from us if waiting for an
incoming call.
This implements all 4 variants the protocol allows for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
AVC-Intra is a range of H.264-compliant intra-only codecs from
Panasonic. The codes and descriptions have been taken from VLC.
The (encumbered) sample I have here produces byte-stream H.264,
including SPS and PPS and no `avcC` box.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3739>
We create a new context in `gst_gl_context_create_thread()` and then
activate it on the current thread. Thereafter we assume that the
current thread continues to be the active thread for that context and
call `gst_gl_context_fill_info()` which asserts that the current
thread is the active thread.
However, if at the same time a different thread calls
`send_message_async()`, it will call into
`gst_gl_window_cocoa_send_message_async()` which will schedule the
message to be invoked using GCD. That anonymous function will also
call `gst_gl_context_activate()`, which creates a race, which can lead
to:
```
gst_gl_context_fill_info: assertion 'context->priv->active_thread == g_thread_self ()' failed
```
Fix it by using `gst_gl_context_thread_add()` to invoke `fill_info()`
on the context.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3732>
The goal of the "global" group-id is to fix new inputs that do not come from the
same "source" as others. In order to ensure all "current" streams have the same
group-id we distribute the first valid group-id to all streams.
This commit fixes two issues with that:
* When inputs are unlinked they weren't always properly resetted (it would only
work if parsebin is used, which is no longer the default in
uridecodebin3/playbin3).
* When computing the global group-id, take into account unset
group-id (i.e. GST_GROUP_ID_INVALID).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1698
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3712>
No matter if they're allocated via GSlice or malloc(). The allocator is
completely irrelevant, all local tags need to be in the primer so they
can be handled.
This didn't have any effect in practice because all local tags that
appear in the muxer are allocated via GSlice. Only from the demuxer they
might be allocated via malloc().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3699>
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>
It might be possible to fulfill those but not with the first caps
structure. Instead of just fixating the first caps structure, check if
the preference can be fulfilled by any of the structures as the first
step.
Without this the following pipeline negotiates to mono after the
decoder because opusenc only has a single channel in its first caps
structure.
gst-launch-1.0 audiotestsrc ! audio/x-raw,channels=2 ! opusenc \
! queue ! opusdec ! queue ! opusenc ! fakesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3689>
When calculating the presentation offset for CMAF input in live
playback, subtract the stream_time of the fragment from the
calculated presentation offset, so that the first fragment
is played at running time zero.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3680>
This should fix pipelines such as this one to work as expected
... ! opusenc ! capsfilter caps='audio/x-opus,
channels=1; audio/x-opus, channels=2' ! ...
The expectation is that the encoder will propose the first structure
before the second one to the source.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3673>
The number of expected pads was:
* Defaulting to 1
* Or being overriden by GST_MESSAGE_STREAMS_SELECTED
This fails if upstream isn't a selectable source and has multiple streams, and
would therefore cause failures with multi-stream gapless playback
Fixes#1672
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3658>
It is quite possible to have the blocking probe called from different streaming
threads when all expected pads are present.
* Notify all waiters by using g_cond_broadcast instead of g_cond_signal
* Properly remove the probe after waiting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3658>
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>
This is recommended by various specifications for such framerates, while
for integer framerates we continue using centiframes to allow for some
more accuracy.
Using N means that no rounding error accumulates, eventually leading to
outputting a packet with a different duration.
Some tools such as MediaInfo determine that a stream is variable
framerate if any packet has a different duration than the others, and
there is no reason I can see for not using the full 4 bytes of
resolution that the mp4 timescale offers.
Example problematic pipeline:
```
videotestsrc num-buffers=5001 ! video/x-raw,framerate=60000/1001,width=320,height=240 ! \
videoconvert ! x264enc bitrate=80000 speed-preset=1 tune=zerolatency ! h264parse ! \
video/x-h264,profile=high-10 ! mp4mux ! filesink location="result2.mp4"
```
This results in a media file that MediaInfo detects as variable
framerate because the 5000th packet has duration 99 instead of 100.
With this patch, the timescale is 60000 and all packets have duration
1001.
Related issue for context: https://bugzilla.gnome.org/show_bug.cgi?id=769041
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3049>
The VAAPI vaQueryVideoProcPipelineCaps() requires the context as the
parameter. So far, we always pass VA_INVALID_ID and it can succeed.
But the API does not say that and in theory, a valid context is required.
Now the new platform really needs a valid context and so we have to
delay that query until the context is created.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3613>
