Call gst_aggregator_selected_samples() after identifying the
caption buffers that will be added as a meta on the next video
buffer.
Implement GstAggregator.peek_next_sample.
Add an example that demonstrates usage of the new API in
combination with the existing buffer-consumed signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1390>
Receiving the WEBKIT_LOAD_COMMITTED event doesn't actually
mean we have committed an SHM buffer / image yet.
As this is the condition we are interested in, check it
instead.
Also wrap g_cond_wait in a loop for extra correctness points.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1476>
"waylandsink: use GstMemory instead of GstBuffer for cache lookup"
changes the cache key to GstMemory, but the cached data still needs
a pointer to the GstBuffer to control the buffer lifecycle.
If the GstMemory used as the cache key moves from one GstBuffer to
another, the pointer in the cached data will be out-of-date.
Update the current GstBuffer pointer for each frame so that it always
represents the currently in use (from attach to release) GstBuffer
for each wl_buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1473>
In case of UWP, documentation from MS is saying that
ActivateAudioInterfaceAsync() method should be called from UI thread.
And the resulting callback might not happen until user interaction
has been made.
So we cannot wait the activation result on constructed() method.
and therefore we should return gst_wasapi2_client_new()
immediately without waiting the result if wasapi2 elements are
running on UWP application.
In addition to async operation fix, this commit includes COM object
reference counting issue around ActivateAudioInterfaceAsync() call.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1466>
Since the commit c29c71ae9d,
device activation method will be called from an internal thread.
A problem is that, CoreApplication::GetCurrentView()
method will return nullptr if it was called from non-UI thread,
and as a result, currently implemented method for accessing ICoreDispatcher
will not work in any case. There seems to be no robust way for
accessing ICoreDispatcher other then setting it by user.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1466>
If the run_async() method is expected to be called from streaming
thread and not from application thread, use INFINITE as timeout value
so that d3d11window can wait UI dispatcher thread in any case.
There is no way to get a robust timeout value from library side.
So the fixed timeout value might not be optimal and therefore
we should avoid it as much as possible.
Rule whether a timeout value can be INFINITE or not is,
* If the waiting can be cancelled by GstBaseSink:unlock(), use INFINITE.
GstD3D11Window:on_resize() is one case for example.
* Otherwise, use timeout value
Some details are, GstBaseSink:start() and GstBaseSink:stop() will be called
when NULL to READY or READY to NULL state change, so there will be no
chance for GstBaseSink:unlock() and GstBaseSink:unlock_stop()
to be called around them. So there is no other way then timeout way.
GstD3D11Window:consturcted() and GstD3D11Window:unprepare() are the case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1461>
This is to avoid having to create a reference chain in decoders from
GstVideoCodecFrame to GstH264PIcture to implementation wrapper.
So this change introduces:
- gst_h264_dpb_remove_outputed (dpb)
- gst_h264_dpb_get_picture(dpb, system_frame_num)
- gst_h264_decoder_get_picture (dec, system_frame_num)
In order to ensure that frames can be looked up during the draining
process, we now first remove all (including reference) frames that
have been outputed but are still in the DPB. Then for each remaining
buffers, we remove it from the DPB to reach reference 1 and output it.
Previously we could take all not outputed outside of the DPB which would
prevent lookup by the base class.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1449>
... when gst_video_decoder_finish_frame() is called.
Some subclasses attach GstBuffer to GstH264Picture as an user data
which will increase reference count of the buffer. It would result
to buffer copy per every frame.
Background here is, GstVideoDecoder wants writable output buffer for
GstMeta handling, and if the output buffer is not writable
(i.e., reference count is not one), the buffer will be copied.
Even if underlying GstMemory wouldn't be copied, buffer copy operation
will introduce extra memory allocation overhead which is not optimal.
By this modification, subclass might be able to receive the last
reference to GstH264Picture
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1449>
All subclasses are retrieving list to get target output frame, which
can be done by baseclass. And pass the ownership of the GstH264Picture
to subclass so that subclass can clear implementation dependent resources
before finishing the frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1449>
While retrieving supported formats by device, the last return might
not be S_OK in case that it's not supported one by us (e.g., H264, JPEG or so).
But if we've found at least one supported raw video format,
we can keep going on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1451>
We don't need to duplicate a method for HRESULT error code to string
conversion. This patch is intended to
* Remove duplicated code
* Ensure FormatMessageW (Unicode version) and avoid FormatMessageA
(ANSI version), as the ANSI format is not portable at all.
Note that if "UNICODE" is not defined, FormatMessageA will be aliased
as FormatMessage by default.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1442>
