When the alignment contains nothing, all its fields are 0 and always
can be satisfied. So there is no need to validate it in this case.
And there are a lot of places just setting this alignment to default
all zero value, this validation generates lots of warnings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4674>
Add d3d11 conversion path to make gst_video_convert_sample() work
for GstD3D11Memory.
Note that just adding "d3d11download" to the exisitng code is
suboptimal from GstD3D11 point of view because:
* d3d11convert element can support crop/colorspace-conversion/scale
all at once while existing software pipeline needs intermediate steps
for the conversion
* "Process everything on GPU then download it to CPU memory" would be likely
faster than "download GPU memory to CPU then processing it on CPU"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2715>
Subclasses may want to override the pad template with different formats
or with a different pad subclass.
The original beahviour is still available by calling
gst_gl_mixer_class_add_rgba_pad_templates() in _class_init() of the
subclass.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4608>
When reconfigure_output_stream entry missing decoder path,
requested_selection should been update with what is really
active/selected immdiately with SELECTION_LOCK hold. So
use an optional message return from reconfigure_output_stream
and post it after release SELECTION_LOCK. This can make sure
other thread call to check_slot_reconfiguration will got
a correct requested_selection.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4599>
If gst_buffer_pool_set_config() fails then the pool will use its old
config. This may include different width or height when
pic_width/pic_height != frame_width/frame_height.
As a result, the assertions in theora_handle_image() will fail.
So check the result of gst_buffer_pool_set_config() and only use the pool
if it succeeds. Otherwise let the parrent decide_allocation() create a new
pool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4600>
If the buffer has no video meta then the meta is created from the local
data. In this case, the other asserts don't actually check anything. So add
another one to ensure that the buffer is actually large enough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4600>
Upon creating a window, glimagesink and osxvideosink now set the policy to
NSApplicationActivationPolicyRegular, which lets us show an icon in the Dock
for convenience and appear in the top menu bar like other apps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4573>
This is no longer needed since the introduction of `gst_macos_main()` in 1.22.
Before that existed, we had a patch for GLib in Cerbero, which did work but made it
impossible to update GLib at all. The code being removed was a fail-safe in case of
running without said patch being applied. It's no longer needed, since for macOS
we just wrap our GStreamer with an NSApplication using `gst_macos_main()`.
Warnings will be displayed if no NSApp/NSRunLoop is found wherever needed,
pointing the user towards using the new API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4366>
Proxy the force-live and min-upstream-latency propertyies to the internal
glvideomixerelement at construction time. force-live has to be set
during construction of the glvideomixerelement, so that has to be
deferred until the _constructed() call. Make sure that all other
existing proxied properties will still get set once the element
is created.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4494>
decodebin3 will do its best to figure out whether a parsebin is required to
process the incoming stream.
The problem is that for push-based stream it could happen that the stream would
not provide any caps, resulting in nothing being linked internally.
Furthermore, there is the possibility that a stream *with* caps would not be
using a TIME segment, which is required for multiqueue to properly work.
In order to fix those two issues, we force the usage of parsebin on push-based
streams:
* When the pad is linked, if upstream can't provide any caps
* When we get a non-TIME segment
Fixes#2521
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4492>
The proxy and queue are created in the gst_gl_window_wayland_egl_open()
function and will be recreated on open. This leaks both objects, the
wayland client documentation mentions that they should be destroyed
using the appropriate destroy functions.
Found during valgrind memory leak testing, these blocks were marked as
definitely lost.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4354>
The first serialized events that can be send on a src pad are a CAPS and then a
SEGMENT event.
When handling events from user in appsrc, we used to send a segment
automatically if the SEGMENT has not been sent yet.
This breaks if the CAPS event was not send either as we were now sending
a SEGMENT before the CAPS.
Fix this by delaying such events until the CAPS has been configured.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4297>
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>
Moving from PLAYING to NULL will set the stop_streaming_threads to TRUE,
but when moving back upwards its not reset to FALSE (as only done in
uncalled init and resume callbacks).
Fix by reseting value in the prepare callback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4309>
Adding propose_allocation is to meet the requirement of Application to
request buffers. Application sometimes need to create buffer pool
and request buffers to maintain buffer management itself, and Gstreamer plugin
import Application's buffers to use. So, add propose_allocation in
appsink like waylandsink and kmssink etc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4185>
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>
This is a follow-up of the previous commit that enabled support for redirection.
The problem is that the urisourcebin that emitted the error redirection never
produced any pads, and therefore was never linked to decodebin3. This resulted
in the code waiting for that (output) item to finally switch over ... which will
never happen.
The fix is done by removing it early if it was never connected to decodebin3.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4252>
The goal of parsebin is to figure out which elements to link together in order
to provide elementary streams given any random input.
The problem is that deciding whether a given stream should still have more
elements plugged in or not was dependent on ... the presence of compatible
decoders (sic).
Instead of that, if we can't plug anymore elements on a given stream *and* it is
detected as being an elementary stream, expose it.
Fixes#2118
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4231>
If sticky events are present on parsebin source pads, we propagate them to the
multiqueue source pads. Those will be propagated on the new urisourcebin source
pads like in the other code paths.
This ensures that STREAM_START event are present on new source pads. If CAPS
event are also present (not guaranteed), they will also be available.
Fixes#2384
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4203>
In gst_video_info_dma_drm_to_caps() the caps are newly created, so there's no
need for make it writable. In gst_video_info_dma_drm_from_caps() a copy of the
caps is done, which implies a gst_caps_make_writable().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4195>
This allow allocating memory from any DRM driver that supports this
method. It additionally allow exporting DMABuf. This allocator depends
on libdrm and will be stubbed if the dependency is missing. This is derived
from kmssink dumb allocator.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3801>
Read and flush console buffer from the console thread immediately,
instead of main thread. Otherwise (if main thread is busy)
the console thread will keep adding idle source and then main thread
will be unresponsive.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4067>
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/4051>
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/3926>
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/3857>
Do not store cached EGL images in GstMemory QData. Instead, use a
per-DmabufUpload GHashTable to store cache entries with a weak
reference to the GstMemory.
This allows two glupload elements on separate tee branches to have
their own EGL image cache. For this pipeline:
gst-launch-1.0 v4l2src ! tee name=t \
t. ! queue ! glupload ! fakesink
t. ! queue ! glupload ! fakesink
this gets rid of the occasional critical error message:
GStreamer-CRITICAL **: 08:26:33.194: gst_mini_object_unref: assertion 'GST_MINI_OBJECT_REFCOUNT_VALUE (mini_object) > 0' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3880>
With the addition of the 'keep-aspect-ratio' sizing policy, content
that doesn't fit the target size is downscaled according to its own
aspect ratio to fit that target size, and centered.
Centering might not always be the desired behaviour, however;
consumers of this API might want to align the resulting picture to
the left or to the right.
To account for any of these cases, add two new properties to the
glvideomixer pad: xalign, and yalign. They operate on normalized
coordinates (0.0 for start, 1.0 for end), and default to 0.5 which
centers content.
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3762>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3762>
The sizing policy allows selecting between the current behavior,
which deforms the texture to fill the width and height of the
pad; and a new 'keep-aspect-ratio' sizing policy, which fits the
texture within the rectangle respecting its original aspect ratio.
The reason for this is that this allows avoiding extra elements
in the pipeline, and reduces the number of buffer passing through
the pipeline.
Most of this code is a direct port of the sizing policy handling
of the compositor element, except it is adapted to operate on GL
texture coordinates through the projection matrix.
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3760>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3760>
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/3757>
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>
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>
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>
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>
Using the "GstBin" flags to check if an adaptive demuxer is streams-aware isn't
a good idea since it prevents using elements which aren't bins.
Instead we see if a collection was posted by the demuxer by the time a pad is
added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3601>
We want to make it so that we prefer a higher, not lower, number of
channels. Otherwise, this pipeline would convert from 2 to 1 channels:
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/3494>
