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>
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>
... if target OS version was specified as Windows 10.
When enabled, desktop application can select target capture
implementation between WinRT and Win32
via GST_USE_MF_WINRT_CAPTURE environment
(e,g., GST_USE_MF_WINRT_CAPTURE=1 for WinRT impl.).
Default is Win32 implementation in case of desktop target.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1434>
Because the valid input formats for screen content coding extension is
a subset of input formats for range extension, user must specify the
profile for screen content coding extension in the caps filter
Example:
gst-launch-1.0 videotestsrc ! video/x-raw,format=NV12 ! msdkh265enc
low-power=1 ! video/x-h265,profile=screen-extended-main ! fakesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1389>
With the asynchronous slice decoding, we only queue up to 2 slices
per frames. That side effect is that now we are dequeuing bitstream
buffers in both decoding and presentation order. This would lead to
a bitstream buffer from a previous frame being dequeued instead of
the expected last slice buffer and lead to us trying to queue an
already queued bitstream buffer.
We now fix this by tracking pending requests. As request are executed
in decoding order, we marking a request done, we can effectively
dequeue bitstream buffer from all previous request, as they have been
executed already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
The decoder is not being access from multiple threads, instead it is
always protected by the streaming lock. For this reason, a
GstAtomicQueue for the request pool is overkill and may even introduce
unneeded overhead. Use a GstQueueArray in replacement, the
GstQueueArray is a good fit since the number of item is predictable and
unlikely to vary at run-time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
In slice mode, we'll do one request per slice. In order to recycle
bitstream buffer, and not run-out, wait for the last pending
request to complete and mark it done.
We only wait after having queued the current slice in order to reduce
that potential driver starvation and maintain performance (using dual
buffering).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
