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>
In fact, all the h264 bit writer have byte aligned output except
the slice header. So we change the API from bit size in unit to
byte size, which is easy to use. For slice header, we add a extra
"trail_bits_num" to return the unaligned bits number.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3193>
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>
Now it uses the JPEG parser in libgstcodecparsers, while the whole
code is simplified by relying more in baseparser class for tag
handling.
The element now signals chroma-format and default framerate is 0/1,
which is for still-images.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1473>
Add gst_h265_parser_identify_and_split_nalu_hevc() method to
handle a case where packetized stream contains start-code prefix.
This new method behaves similar to exisiting gst_h265_parser_identify_nalu_hevc()
but it will scan start-code prefix to split given data into
NAL units.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2394>
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.
While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
If things progress fast enough, some state changes may not be seen be
the waiting code.
Fix by:
1. keeping a list of all the state changes
2. waiting checks each entry and if the relevant state is found, all
states up to and including then are removed.
This ensures that any waits will see all the state sets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
Input (sink pads) is the already-ssrc-muxed stream with the relevant rtp
sdes header extensions already applied:
- mid
- stream-id
- repaired-stream-id
Output (src pads) have the pads separated into individual ssrc's as
that's what rtpbin gives us.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
When creating a transceiver when creating an answer, the media kind of the
transceiver was never set correctly initially. This would lead to a
GST_WARNING being produced about changin a transceiver's media kind.
Fix by retrieving the GstSDPMedia kind from the offer instead as the answer
GstSDPMedia has not been set as the answer caps have not been chosen yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
