As specified in EIA/CEA-608-B section 8.4:
When closed captioning is used on line 21, field 2, it shall conform
to all of the applicable specifications and recommended practices as
defined for field 1 services with the following differences:
a) The non-printing character of the miscellaneous control-character pairs
that fall in the range of 14h, 20h to 14h, 2Fh in field 1, shall be replaced
with 15h, 20h to 15h, 2Fh when used in field 2.
b) The non-printing character of the miscellaneous control-character pairs
that fall in the range of 1Ch, 20h to 1Ch, 2Fh in field 1, shall be replaced
with 1Dh, 20h to 1Dh, 2Fh when used in field 2.
This means simply switching the "field" field in the caps isn't enough for
converting raw 608 from one field to another, some control codes also
need to be amended.
+ Adds simple test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4126>
GstBufferPool implementation was referenced for this GstD3D11PoolAllocator,
for example GstAtomicQueue, various atomic operations, and GstPoll ones.
However, such combination seems to be almost pointless
since gst_poll_{read,write}_control() takes mutex and also
GstPoll uses Win32 event handle internally.
Use simple SRWLOCK and CONDITION_VARIABLE instead, and don't make things
complicated/inefficient.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2887>
The code wants to prepend one byte to every byte pair. It correctly did
so by working backwards pair-wise, but then didn't work backwards
instead of each individual pair / future triplet, overwriting
information before attempting to read it.
The code also failed to update the len pointer after prepending.
This fixes both issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4100>
Removing sockets from the epoll for cancellation is unreliable and might
not be thread-safe. Rather, have SRT watch a FD from the cancellable if
available. Keep the cancellable cancelled while we're not open.
Use the regular single-socket `sock` and `poll_id` fields for the
listening thread instead of duplicating them.
Before polling we need to check the socket state. SRT closes broken
sockets by itself and when the epoll contains our cancellation FD it can
no longer be empty, which was an error before.
Treat more failures in the read and write operations as an opportunity
to try a reconnect.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4087>
Seems that SRT can remove the socket from the poll by itself when the
connection gets closed. Consider this an error condition and ensure we
only "abort successfully" when we're actually trying to unlock.
Needs more investigation but this is enough to prevent the element from
getting stuck not reporting an error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4087>
Recursively invoking the NSMainLoop can cause crashes in
applications that don't expect it. Instead of waiting for
permission to be granted, move the wait later - until we
actually need device permissions when starting the capture
session. That moves the wait into the streaming thread
instead of the application thread that's setting the pipeline
state to READY.
Instead of a manual state change implementation to open
and close the device, use the basesrc start/stop methods that
are intended for the purpose.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4083>
There may be garbage or some bits before a SOI comes in some problematic
mjpeg streams. For example, some network error may cause the EOI marker
of the previous frame lost, and when the new frame's SOI comes, we still
use the state of the last frame, which will generate errors.
For this kind of frames without EOI, if that frame already has some data
(the SOS segment is detected), we still push it as a frame with CORRUPTED
flag set. But if not, we just discard all the data before the new SOI.
Co-Authored-By: Víctor Jáquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4039>
The previous implementation was a bit primitive, assuming the subclass
had registered a template name starting with sink_ . Instead make
the effort of parsing the actual template name, and use that to generate
the final pad name.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4032>
These checks were introduced to prevent exposing ARGB64/RGBA64 in the caps
when running on M1 Pro/Max with macOS <13 because of a bug in VideoToolbox.
Unfortunately, the initial buffer size of 15 is too short when running
in a VM - the CPU brand string there looks like "Apple M1 Pro (Virtual)",
which due to its length causes sysctlbyname to return -1, resulting in
broken formats still showing up in the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4080>
We did several things to enable the new memory logic in msdkdec:
(1) We always use video memory for decoder in linux path;
(2) We give negotiated pool to alloc_pool stored in GstMsdkContext which
will be used in callback mfxFrameAllocator:Alloc to alloc surfaces as
MediaSDK needs, and this pool is also available for decoder itself;
(3) We modify decide_allocation process, that is we make pool negotiaion
before gst_msdk_init_decoder to ensure the pool is decided and ready for
use in mfxFrameAllocator:Alloc callback; then we will consider the case
when we need to do the gpu to cpu copy.
(4) In gst_msdkdec_finish_task, we modify the way for copy following the
logic in (3).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3439>
Add a pool creation function name as 2 for later use which will create
va pool for video memory in linux and keep system pool for windows.
This gst_msdkdec_create_buffer_pool2 will replace gst_msdkdec_create_buffer_pool
when all the memory allocation modifications are ready in the commits after.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3439>
Rewrite gst_msdk_frame_alloc and name it as xxx_2 before applying it.
It uses negotiated bufferpool stored in GstMsdkContext to allocate buffers
in the callback MfxFrameAllocator:Alloc, then extract VASurface from buffer,
wrap it as mfxMemIDs and pass these IDs to MediaSDK/oneVPL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3439>
The `add_candidate` vfunc of the GstWebRTCICE interface gained a GstPromise
argument, which is an ABI break. We're not aware of any external user of this
interface yet so we think it's OK.
This change is useful in cases where the application needs to bubble up errors
from the underlying ICE agent, for instance when the agent was given an invalid
ICE candidate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3960>
The signal triggers an asynchronous task on the PC thread but in some cases it
can be useful for apps to be notified when the task completed. This method of
the PeerConnection spec also returns a Promise so the interface is now more
coherent with the spec.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3960>
