Interlaced MJPEG is a big hack. Most of the streams we've found are from old
AVID tools. There are two methods to detect interlaced stream: the container
offers a height bigger (or double) than the image's height in SOF. The other
is from a APP0 marker.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5838>
Serialize every GstMeta that supports serialization into the NEW_BUFFER
payload. This is especially important for GstVideoMeta in the case of
multiplanar buffers, or if stride!=width.
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5355>
A payload of 0x80 0x80 means that it's padding. It's not a good idea to
throw this away though, because of the cc_valid field.
According to CEA 10-B section 25.2.1, if cc_valid is zero, the run-in
clock and start bit should not be generated. In practice, this means
that any closed captions will be erased and the end-user TV will show
that captions are not available for this stream. This might have
undesired consequences, e.g. we were just showing a long line of
captions and we disable it before the user has had time to read it, or
you can't enable closed captions during silence/music intervals.
We cannot reliably detect whether there's a currently-silent closed
caption stream or just nothing, but we have this information coming from
upstream, so we can at least not discard it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5508>
This pair of elements, inspired from shmsink/shmsrc, send unix file
descriptors (e.g. memfd, dmabuf) from one sink to multiple source
elements in other processes.
The unixfdsink proposes a memfd/shm allocator, which causes for example
videotestsrc to write directly into memories that can be transfered to
other processes without copying.
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5328>
Depending on the exact output format, 0x00 may be a better default for
padding than 0x80. 0x00 is the recommended padding value when used in
CDP (and cc_data) but is not when used in s334-1a. See CTA-708-E 4.3.5
amd SMPTE 334-1-2007 5.3.2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4578>
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>
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>
Raw 608 caps can now contain a "field" field. On the input side it
signifies that the input raw 608 is attached to either field 0 or 1,
on the output side it allows selecting whether to extract the raw 608
data for field 0 or 1 for field-aware formats.
In addition, it is also allowed to use ccconverter to "convert" 608
field 0 to 608 field 1 (and conversely), this is passthrough as the
change only needs to happen in the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4031>
If an input is malformed (only produces cea608 field 1 cc_data) then
when in passthrough we would effectively be dropping every second cea608
on output as we would not store any unused cea608 data.
Fix by having all code paths go through the framerate conversion code
which will store and retrieve any relevant data across buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.
Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
doesn't align on 20 millisecond frame size.
The AMR-WB codec imposes a fixed 20 millisecond frame size. In its current
form, the `voamrwbenc` plugin deals with this limitation by discarding any
audio at the end of the stream that falls short of 20 milliseconds. This patch
keeps the audio data, and appends silence to the end to preserve frame size
alignment.
The patch also adds tests to check for the updated behavior. I noticed that
tests weren't being built, so I changed the build to allow for building the
tests when the `tests` and `voamrwbenc` options are set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3027>
This avoids getting in a bunch of corner cases. We'd have to insert
a "rejected" line from the start as a place-holder to get around this,
but the rest of the code just becomes more complicated, so just
disallow it for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
There might be a sequence of event and buffer flow:
- Got stream-start/caps/segment events
- Got flush events
- And then buffers with a new segment event
In the above case, stream-start and caps event might not be reached to
peer proxysrc if peer proxysrc is not ready to receive them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1552>
This new signal allows data-channel consumers to configure signal handlers on a
newly created data-channel, before any data or state change has been notified.
The webrtcin unit-tests were refactored to make use of this new signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2427>
WHen bundling, if multiple medias are used with the same media payload, then
each of the fec/rtx/red additions would add a distinct payload. This could
very easily overflow the available payload space.
Instead, track the relationship between the media payload value and
the relevant fec/rtx/red payload values and reuse them whenever
necessary, even when bundling.
e.g.
...
a=group:BUNDLE video0 video1
m=video 9 UDP/SAVPF 96 97
a=mid:video0
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...
m=video 9 UDP/SAVPF 96 97
a=mid:video1
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2474>
