We can only convert from non-cdp to cdp within the confines of valid cdp
framerates. The existing caps negotiation code was allowing any
framerate to convert to a cdp output which is incorrect and would hit an
assertion later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2372>
By this commit, following formats will be newly supported by d3d11 elements
* Y444_{8, 12, 16}LE formats:
Similar to other planar formats. Such Y444 variants are not supported
by Direct3D11 natively, but we can simply map each plane by
using R8 and/or R16 texture.
* P012_LE:
It is not different from P016_LE, but defining P012 and P016 separately
for more explicit signalling. Note that DXVA uses P016 texture
for 12bits encoded bitstreams.
* GRAY:
This format is required for some codecs (e.g., AV1) if monochrome
is supported
* 4:2:0 planar 12bits (I420_12LE) and 4:2:2 planar 8, 10, 12bits
formats (Y42B, I422_10LE, and I422_12LE)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2346>
This is for the case where there answerer forces a specific media type
for a m-line, but he origin offer only has the other media type. In this
case, we will create a second transceiver on receiving the offer and add
the desired media type using renegotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
Offset and size are stored as 32 bit guint and might overflow when
adding the nal_length_size, so let's avoid that.
For the size this would happen if the AVC/HEVC NAL unit size happens to
be stored in 4 bytes and is 4294967292 or higher, which is likely
corrupted data anyway.
For the offset this is something for the caller of these functions to
take care of but is unlikely to happen as it would require parsing on a
>4GB buffer.
Allowing these overflows causes all kinds of follow-up bugs in the
h2645parse elements, ranging from infinite loops and memory leaks to
potential memory corruptions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2103>
Prior to that, cccombiner's behaviour was essentially that of
a funnel: it strictly looked at input timestamps to associate
together video and caption buffers.
This patch instead exposes a "schedule" property, with a default
of TRUE, to control whether caption buffers should be smoothly
scheduled, in order to have exactly one per output video buffer.
This can involve rewriting input captions, for example when the
input is CDP sequence counters are rewritten, time codes are dropped
and potentially re-injected if the input video frame had a time code
meta.
Caption buffers may also get split up in order to assign captions to
the correct field when the input is interlaced.
This can also imply that the input will drift from synchronization,
when there isn't enough padding in the input stream to catch up. In
that case the element will start dropping old caption buffers once
the number of buffers in its internal queue reaches a certain limit
(configurable).
The property is exposed so that existing users of cccombiner can
revert back to the original behaviour, but should eventually be
removed, as that behaviour was simply inadequate.
This commit also disallows changing the input caption type, as
this would needlessly complicate implementation, and removes
the corresponding test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2076>
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>
It was possible to generate a SDP that had an RTX payload type
that matched one of the media payload types when providing caps via
codec_preferences without any sink pads.
Fixes
m=video 9 UDP/TLS/RTP/SAVPF 96
...
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack pli
a=fmtp:96 apt=96
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
