Before trying to retrieve a GMainContext from a provided
GstPlayerSignalDispatcher, check that it is actually
GstPlayerGMainContextSignalDispatcher. If not, use the
default GMainContext for dispatching signals via the adapter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7497>
This mirrors the behaviour in vp8enc / vp9enc and is generally more
useful than using any framerate from the caps as it provides some degree
of accuracy if the stream doesn't have timestamps perfectly according to
the framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6922>
It might happen that the key event arrives when the d3d11videosink
is stopping. In case of GstD3D11WindowWin32 it can raise a
navigation event even when the sink is already freed, because the
window object's refcount may reach 0 in the window thread. In
other words sometimes the GstD3D11WindowWin32 lives few ms more
then the GstD3D11VideoSink, because it's freed asynchronously.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6493>
In the situation where playback starts from a keyframe before
the target playback segment, then the first buffers will be
outside the configured segment and gst_segment_to_stream_time()
will return GST_CLOCK_TIME_NONE unconditionally.
If drop-out-of-segment is false, the RTP buffers will not be
dropped, but will be sent witout ONVIF extension timestamps
and given GST_CLOCK_TIME_NONE timestamps on the receiver.
Instead, use gst_segment_to_stream_time_full() to extrapolate
stream time outside the segment so that such buffers still
get assigned their correct timestamps on the receiver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6287>
Don't accidentally include the stuffing byte (if present)
into the bottom field size. It should only be included in the
total segment length.
Fixes problems with FFmpeg not rendering the subtitles
with a stuffing byte, giving a "Invalid object location!" error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6281>
Unprepare method posts WM_GST_D3D11_DESTROY_INTERNAL_WINDOW
command to the window queue, and from that moment considers
internal_hwnd to be released, and so it sets it to null.
The problem is that it's possible that right at that moment
the window thread might be already processing some other
command, or just another command might be already in the queue.
On practice we met a crash when WM_PAINT got processed in between
(unprepare already finished and WM_GST_D3D11_DESTROY_INTERNAL_WINDOW
was not handled yet)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6230>
The original idea was to select the type of mapping (either using derive images
or downloading the image) in runtime, under the assumption that both methods
shared the same memory layout (offsets and strides), because a single
GstVideoMeta is assigned by the buffer pool at allocation time. Nonetheless, in
recent hardware this assumption is invalid, raising memory access errors.
This patch removes completely the mapping type selection at runtime, using the
method selected when the allocator is configured, synced with the bufferpool
allocation.
This problem was fixed originally for iHD driver only. But now it makes sense to
remove all of it.
Original-patch-by: Mengkejiergeli Ba <mengkejiergeli.ba@intel.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6204>
This reverts questionable commit 009bc15f33
which looks completely wrong.
The GstWasapi2RingBuffer:buffer_size variable is used to
calculate available buffer size we can write
(i.e., available size = buffer_size - padding_size).
But the commit makes the size to be exactly same as buffer period.
Then, it can confuse this element as if the endpoint buffer is full on
I/O event callback (if padding size is equal to buffer period)
but it's not true.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2870
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6140>
WebKit commit b12e7ed2ad3a ("[WPE] Upstream the new WPE platform API
https://bugs.webkit.org/show_bug.cgi?id=265286")[1] added a `WPEView` typedef
which clashes with our `WPEView` class.
Rename the `WPEView` class to `GstWPEThreadedView` to avoid the collision.
Also prefix the `WPEContextThread` class with `Gst` and rename the
source files to reflect the new class name and use lowercase while at it
for consistency
[1] b12e7ed2ad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6096>
* Bump the rank of the musepack v7/v8 FFmpeg demuxers to SECONDARY
* Bump the rank of the musepack v7/v8 FFmpeg audio decoders to SECONDARY
* Demote the rank of the musepackdec element to MARGINAL
This is for two reasons:
* The musepack library is no longer maintained, whereas the FFmpeg
implementation can/will receive fixes
* The `musepackdec` implementation was a all-in-one "parsing and decoding" blob
which doesn't play nicely with decodebin3 and others
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3033
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6085>
Remove optional sprop-stereo and sprop-maxcapture fields from Opus
remote offer caps before intersecting with local codec preferences.
According to https://datatracker.ietf.org/doc/html/rfc7587#section-7.1
those fields are sender-only informative, and don't affect
interoperability.
Fixes cases where the webrtc media will end up receive-only if the
local side wants to send stereo but the remote is sending mono, or
vice versa.
There may be other fields in other codecs, so the implementation
anticipates needing to add further fields and codecs in the future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5999>
