When carrying over existing GstStream to a new GstStreamCollection we need to
check whether they *actually* were being used in the previous collection.
This avoids adding unknown streams (metadata, PSI, etc...) to the collection on
updates.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
Instead of implementing exactly the same thing ourself but making
`GstBus` not know that it is the case.
Since we are *sure* that the bus can't have been access at the point
where we add the watch we are guaranteed that the current thread
maincontext is going to be used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1870>
Otherwise, when rtpm2src cancels an inflight operation that has a queued
message stored, then the rtmp connection operation is not stopped.
If the cancellation occurs during rtmp connection start up, then
rtpm2src does not have any way of accessing the connection object as it
has not been returned yet. As a result, rtpm2src will cancel, the
connection will still be processing things and the
GMainContext/GMainLoop associated with the outstanding operation will be
destroyed. All outstanding operations and the rtmpconnection object will
therefore be leaked in this case.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1425
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1862>
This function takes the sock lock. This can result in a deadlock when
another thread holding the sock lock is trying to take the object lock.
Thread A (Holds object lock, wants sock lock):
#2 gst_srt_object_get_stats at gst-plugins-bad/ext/srt/gstsrtobject.c:1753
#3 gst_srt_object_get_property_helper at gst-plugins-bad/ext/srt/gstsrtobject.c:409
#4 gst_srt_sink_get_property at gst-plugins-bad/ext/srt/gstsrtsink.c:95
#5 g_object_get_property from libgobject-2.0.so.0
Thread B (Holds sock lock, wants object lock):
#2 gst_element_post_message_default at gstreamer/gst/gstelement.c:2069
#3 gst_element_post_message at gstreamer/gst/gstelement.c:2123
#4 gst_element_message_full_with_details at gstreamer/gst/gstelement.c:2259
#5 gst_element_message_full at gstreamer/gst/gstelement.c:2298
#6 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1407
#7 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#8 gst_srt_object_write_to_callers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#9 gst_srt_object_write at gst-plugins-bad/ext/srt/gstsrtobject.c:1598
#10 gst_srt_sink_render at gst-plugins-bad/ext/srt/gstsrtsink.c:179
Fixes d2d00e07ac.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1861>
Various software, including ffmpeg's Decklink support, fails parsing CDP
packets that contain anything but CC data in the CDP packets.
Based on this property, timecodes are not written into the CDP packets
even if they're present.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
Add a new property "render-stats" to allow rendering statistics
data on window for debugging and/or development purpose.
Text rendering will be accelerated by GPU since this implementation
uses Direct2D/DirectWrite API and Direct3D inter-op for minimal overhead.
Specifically, text data will be rendered on swapchain backbuffer
directly without any copy/allocation of extra texture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1830>
Context creation and retrieval is required, the symbols are exported
with the header missing. Users most likely define GST_USE_UNSTABLE_API
so they're aware of the implications of using a header that might change
between releases.
Signed-off-by: Marius Vlad <marius.vlad@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1688>
Instead of waiting so that we can simply use a clocksync element as
filter, otherwise we won't know the pipeline is live as it won't
return NO_PREROLL as one would expect in that case.
Adding it right away shouldn't create any issue, both ways are fine.
Since GstVaDecodePicture is destroyed completely with its free() function and
it's used as destroy notify by codecs picture, there's no need to call
gst_va_decoder_destroy_buffers() externally, since the codecs base classes
destroy the codec picture when it's required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1841>
The current way of GstVaDecodePicture's finalize will leak some
resource such as parameter buffers and slice data.
The current way deliberately leaves these resource releasing logic
to va decoder related function and trigger a warning if we free the
GstVaDecodePicture without releasing these resources.
But in practice, sometimes, you do not have the chance to release
these resource before picture is freed. For example, H264/Mpeg2
support multi slice NALs/Packets for one frame. It is possible that
we already succeed to parse and generate the first several slices
data by _decode_slice(), but then we get a wrong slice NAL/packet
and fail to parse it. We decide to discard the whole frame in the
decoder's base class, it just free the current picture and does not
trigger sub class's function again. In this kind of cases, we do
not have the chance to cleanup the resource, and the resource will
be leaked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1841>
The base class is now a bin which wraps the `overlaycomposition`
element and implements the `draw` signal.
This way we support all the video formats the GstVideoOverlayComposition
API supports and the blending code can be reused. It is also possible
to have the blending happen in the sinks now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1829>
Even if resolution and/or bitdepth is not updated, required
DPB size can be changed per SPS update and it could be even
larger than previously configured size of DPB. If so, we need
to reconfigure DPB d3d11 texture pool again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1839>
In order to honor GST_BUFFER_POOL_ACQUIRE_FLAG_DONTWAIT in VA pool, allocators'
wait_for_memory() has to be decoupled from their prepare_buffer() so it could be
called in pools' acquire_buffer() if the flag is not set.
wait_for_memory() functions are blocking so the received memories are assigned
to the fist requested buffer, if multithreaded calls. For this a new mutex were
added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
An application, using for example appsink, can hold buffers from any
va allocator after setting the pipeline to NULL. We need to destroy
the allocator when that memory is unrefed.
This patch juggles a bit with the allocator reference count in
memories in order to achieve this:
1. When memory is created no alloc ref is modified
2. When memory is released, alloc ref is decreased
3. When memory is reassiged to a buffer, alloc ref is increased
4. When memory is flushed, alloc ref is increased becase it is going
to be decreased in gst_memory_unref()
Also this patch moves the deallocation of member variables to
finalize() rather than dispose()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
