Some functions now accept a generic 'gpointer data' parameter because
they can work either on a single buffer or a buffer list.
However the comments were still referring to the old 'GstBuffer *buffer'
parameter, so update the comments to match the actual functions
signature.
So far we assumed that if all sources are bye, this meant we needed to
send an EOS on the RTCP sink. The problem is that this case may happens
if we only had one internal source and it detected a collision.
So now we limit the EOS forwarding to when there is a send_rtp_sink pad
and that this pad has received EOS. We don'tcheck the recv_rtp_sink
since the code does not wait for the bye to be send before sending EOS
to the RTCP src pad.
RF64 encode support was added to wavenc quite some time
ago, but not declared in wavparse. It seems wavparse can
decode it though, so add it to the sink pad.
The RF64 support was added in
https://bugzilla.gnome.org/show_bug.cgi?id=735627
This is useful when implementing custom retransmission mechanism like
RIST to prevent RTCP from being produces for the retransmitted SSRC.
This would also be used in general for various purpose when customizing
an RTP base pipeline.
If it goes over 2^15 packets, it will think it has rolled over
and start dropping all packets. So make sure the seqnum distance is not too big.
But let's not limit it to a number that is too small to avoid emptying it
needlessly if there is a spurious huge sequence number, let's allow at
least 10k packets in any case.
Recent changes in ccextractor were attaching timecode meta to the closed
caption track. We shouldn't write timecode information for the closed
caption trak.
And let it the oportunity to get its other pad linked
Example:
```
$ gst-launch-1.0 uridecodebin uri=file:///home/thiblahute/gst-validate.save/gst-integration-testsuites/testsuites/../medias/defaults/flv/819290236.flv caps=audio/x-raw expose-all-streams=FALSE ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Internal data stream error.
Additional debug info:
../subprojects/gst-plugins-good/gst/flv/gstflvdemux.c(2760): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0:
streaming stopped, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
```
Modify the caps string to allow width and height greater than 4096.
There is no need to restrict it since the matroska format allows the
width and height values to be up to eight bytes long, and this also
applies to the webm subset of the format.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/550
When multiple nals are aggrgated, the marker bit should be associated only
with the last NAL of the packet. Otherwise we may break rendering in with
AU alignment.
And also add a property for setting this. By default it has the same
value as the metadatacreator metadata.
Various software is using encoder instead of metadatacreator, others are
using them both for different purposes. As such it's useful to have
support for setting both here.
Blocking in change_state() is a recipe for disaster, even more so if
we wait for another thread that also calls into various element API and
could then lead to deadlocks on e.g. the state lock.
EA608 closed caption tracks are a bit special in that each sample
can contain CCs for multiple frames, and CCs can be omitted and have to
be inferred from the duration of the sample then.
As such we take the framerate from the (first) video track here for
CEA608 as there must be one CC byte pair for every video frame
according to the spec.
For CEA708 all is fine and there is one sample per frame.
The duration field being a uint64, is stored in 8 bytes, not 4. So the offset of
the following field, language code, needs to be updated accordingly so that the
parsed language code is not garbage.
The documentation of "port-range" implies that passing NULL should be
valid, but currently it is not. Without this check, the sscanf() call
will crash.
This reverts commit dcd3ce9751.
This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.
This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.
Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.
Fixes#537
This macro is not longer used. It was secretly checking if that nal was
a slice, and confusingly name to that one may think it was checking if
the nal is an AUD.
The code was reading the timestamp from the adapter before pushing the
new buffer into it. As a side effect, if the adapter was empty, we'd end
up using an older timestamp. In alignment=au, it means that all
timestamp was likely one frame in the past, while in alignment=nal, with
multiple slices per frame, the first slice would have the timestamp of
the previous one.
