The converter might have a non-passthrough mix-matrix. The converter
can determine whether it should pass through, so let it, then remove it
if it's indeed a passthrough.
FIXME: Not converting when we need to but the config is invalid (e.g.
because the mix-matrix is not the right size) produces garbage. An
invalid config should cause a GST_FLOW_NOT_NEGOTIATED.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1070>
The users of this API need to be able to differentiate between EINTR
and ERROR. For example, in rtspsrc, gst_rtsp_conninfo_connect()
behaves differently when gst_rtsp_connection_connect_with_response_usec()
returns an ERROR or EINTR. The former is an element error while the
latter is simple a GST_ERROR since it was a user cancellation of the
connection attempt.
Due to this, rtspsrc was incorrectly emitting element errors while
going to NULL, which would or would not reach the application in
a racy manner.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1069>
For reverse playback, we are always copying decoded
frame to downstream buffer. So the pool size can be
and need to be large enough.
In case that forward playback, however, we need to restrict
the max pool size for performance reason. Otherwise decoder
will keep copying decoded texture to downstream buffer pool
if decoding is faster than downstream throughput
performance and also there are queue element between them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2083>
The old code had a couple of issues that all lead to potential memory
safety bugs.
- Use a constant for the Wavpack4Header size instead of using sizeof.
It's written out into the data and not from the struct and who knows
what special alignment/padding requirements some C compilers have.
- gst_buffer_set_size() does not realloc the buffer when setting a
bigger size than allocated, it only allows growing up to the maximum
allocated size. Instead use a GstAdapter to collect all the blocks
and take out everything at once in the end.
- Check that enough data is actually available in the input and
otherwise handle it an error in all cases instead of silently
ignoring it.
Among other things this fixes out of bounds writes because the code
assumed gst_buffer_set_size() can grow the buffer and simply wrote after
the end of the buffer.
Thanks to Natalie Silvanovich for reporting.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/859
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/902>
Decoder might be able to copy decoded texture to the other buffer pool
during playback depending on context. In that case, copied one
has no D3D11_BIND_DECODER bind flag.
If we used ID3D11VideoProcessor previously for decoder texture,
and incoming texture supports ID3D11VideoProcessor as well even if it has no
D3D11_BIND_DECODER flag (having D3D11_BIND_RENDER_TARGET for example),
allow zero-copying instead of using our fallback texture.
Frequent conversion tool change (between ID3D11VideoProcessor and generic shader)
might result in inconsistent image quality.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2084>
This usually doesn't matter, but it is disruptive when streaming from
a shared media since it will pause all other clients when any client
exits.
This new behaviour is opt-in and should be safe because you need to
set the NULL state on rtspsrc directly, instead of just on the
pipeline. See the updated documentation for an explanation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/901>
... instead of QueryInterface-ing per elements. Note that
ID3D11VideoDevice and ID3D11VideoContext objects might not be available
if device doesn't support video interface.
So GstD3D11Device object will create those objects only when requested.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2079>
Direct3D11 objects are COM, and most COM C APIs are verbose
(C++ is a little better). So, by using C++ APIs, we can make code
shorter and more readable.
Moreover, "ComPtr" helper class (which is C++ only) can be
utilized, that is very helpful for avoiding error-prone COM refcounting
issue/leak.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2077>
While preparing a blist for pushing, some RTP header extension may
request caps change for a specific buffer in the list. When this
happens, depayloader should immediately push those buffers from the list
that precede the currently processed buffer (for which the caps change
was requested) and only then apply the new caps to the src pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
Some header extensions may need to read information from the payloader's
sink caps. Introduce gst_rtp_header_extension_update_from_sinkcaps ()
that passes the caps to the extension, which can then use it to update
its internal state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
If the driver supports it (iHD, so far) and the parameter -d is set,
the direction of the video will be changed randomly.
In the code you can select, at compilation time, if the direction
change is done by element's property or by pipeline events.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2074>
Added helper function _update_passthrough() which will define and set
the pass-through mode of the filter, and it'll either reconfigure both
pads or it will just mark the src pad for renegotiation or nothing at
all.
There are cases where both pads have to be reconfigured (direction
changed, for example), other when just src pad has to (filters
updated) or none (changing to ready state).
The requirement of renegotiation depends on the need to enable/disable
its VA buffer pools.
This patch sets pass-through mode by default, so the buffer pools
aren't allocated if no filtering/direction operations are defined,
which is the correct behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2074>
This because underlying driver may have constraint on
buffer size to be dependent on period size, so period
time needs to be set first.
For e.g. Xilinx ASoC driver requires
buffer size to be multiple of period size for it's DMA
operation.
alsa-utils also set period time first as seen in below commit :
9b621eeac4
Tested it on zcu106 board with HDMI based record and playback.
Also tested on Intel PC using Logitech C920 Webcam mic and ALC887-VD
Analog for playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1040>
