SRT provides the original timestamp of a packet (with drift/skew corrected for
local clock), which is what should be used for timestamping the outgoing
buffers. This ensures that we output the packets with the same timestamp (and by
extension rate) as the original feed.
Also detect if packets were dropped (by checking the sequence number) and
properly set DISCONT flag on the outgoing buffer.
Finally answer the latency queries
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1658>
Updated Decklink SDK to version 11.2 in order to support newer cards like the Decklink 8K Pro.
This required to replace the duplex property by a profile property.
Profile values can be the following:
- bmdProfileOneSubDeviceFullDuplex
- bmdProfileOneSubDeviceHalfDuplex
- bmdProfileTwoSubDevicesFullDuplex
- bmdProfileTwoSubDevicesHalfDuplex
- bmdProfileFourSubDevicesHalfDuplex
Fixes#987
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
1. Allocators don't implement memory free() methods since all the memories will
implement dispose() returning FALSE
2. Memory/miniobject dispose() will act as memory release, enqueueing the
release memory
3. A new allocator's method prepare_buffer() which queries the released memory
queue and will add the requiered memories to the buffer.
4. Allocators added a GCond to synchronize dispose() and prepare_buffer()
5. A new allocator's method flush() which will free for real the memories.
While the bufferpool will
1. Remove all the memories at reset_buffer()
2. Implement acquire_buffer() calling allocator's prepare_buffer()
3. Implement flush_start() calling allocator's flush()
4. start() is disabled since it pre-allocs buffers but also calls
our reset_buffer() which will drop the memories and later the
buffers are ditched, something we don't want. This approach avoids
buffer pre-allocation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
Renamed the first variable member of GstVaMemory from parent to mem in
order to avoid confusion with GstMemory's parent.
When freeing the structure, memory's parent is check in order to
decide if surfaces has to be destroyed or not, since only the parent
class have to destroy it.
Removed GST_MEMORY_FLAG_NO_SHARE in memory initialization, since it is
deprecated.
Implemented allocator's share virtual method which creates a new
shallow GstVaMemory structure based on the passed one which will be
it's parent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
Staging texture is used for memory transfer between system and
gpu memory. Apart from d3d11{upload,download} elements, however,
it should happen very rarely.
Before this commit, d3d11bufferpool was allocating at least one
staging texture in order to calculate cpu accessible memory size,
and it wasn't freed for later use of the texture unconditionally.
But it will increase system memory usage. Although GstD3D11memory
object is implemented so that support CPU access, most memory
transfer will happen in d3d11{upload,download} elements.
By this commit, the initial staging texture will be freed immediately
once cpu accessible memory size is calculated.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1627>
These function were repeated in the different implemented
elements. This patch centralize them.
The side effect is dmabuf memory type is no longer checked with the
current VAContext, but assuming that dmabuf is a consequence of caps
negotiation from dynamic generated caps templates, where the context's
memory types are validated, there's no need to validate them twice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1644>
An oddness of wasapi loopback feature is that capture client will not
produce any data if there's no outputting sound to corresponding
render client. In other words, if there's no sound to render,
capture task will stall. As an option to solve such issue, we can
add timeout to wake up from capture thread if there's no incoming data
within given time interval. But it seems to be glitch prone.
Another approach is that we can keep pushing silence data into
render client so that capture client can keep capturing data
(even if it's just silence).
This patch will choose the latter one because it's more straightforward
way and it's likely produce glitchless sound than former approach.
A bonus point of this approach is that loopback capture on Windows7/8
will work with this patch. Note that there's an OS bug prior to Windows10
when loopback capture client is running with event-driven mode.
To work around the bug, event signalling should be handled manually
for read thread to wake up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1588>
In live streaming, buffers sent by souphttpsrc are pushed to the live
adapter. The buffers in the adapter are sent out of mssdemux when it
is greater than 4096 bytes.
Occasionally, when seeking in live streams, if seek occurs just
after the last data chunk was received, and if this data chunk is
smaller than 4096 bytes, it will be kept in the live adapter.
This remaining data in the live adapter will be erroneously prepended
to the new data that is downloaded after seek and pushed out.
When qtdemux receives this data, since it does not start with
a moof box, it is impossible to demux the fragment, and bogus
size error will occur.
Clear out the live adapter on seek so that no unnecessary remaining
data is pushed out together with the new fragment after seeking.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1345>
