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.
The marker bit is used for efficient decoding. The assumption that
it should be set on the AUD is wrong, since the AUD is conceptually
starts the frame, while the marker is to indicate the end.
So properly set the marker bit as soon as we know we are ending an
AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
flag.
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.
The marker bit is used for efficient decoding. The assumption that
it should be set on the AUD is wrong, since the AUD is conceptually
starts the frame, while the marker is to indicate the end.
So properly set the marker bit as soon as we know we are ending an
AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
flag.
Don't allow external encoder to use one of the reserved NAL type
implicated in NAL aggreation. These out-of-spec NAL types, if passed
from the outside world will lead to an invalid RTP payload being
created.
When the EOS event is received, run all timers immediately and avoid
pushing the EOS downstream before this has been run. This ensures that
the lost packet statistics are accurate.
After EOS is received, it is pointless to wait for further events,
specially waiting on timers. This patches fixes two cases where we could
wait instead of returning GST_FLOW_EOS and trigger a spin of the loop
function when EOS is queued, regardless if this EOS is the queue head or
not.
stream.segment should be updated with the values of the current edit
list, also when a new `moov` is received. Unfortunately this was not
being the case because of an early return.
As a consequence of this bugs, no end of movie clipping was being
performed on the new moov and no segment event was being emitted.
When performing stream switching (e.g. in MSE) the new moov may have a
different edit list. This is often the case when switching between
baseline H.264 (which lacks B-frames) and more demanding profiles. For
this reason it's important to emit a new segment in order to be able
to get matching stream times.
This patch moves the initialization of QtDemuxStream.segment from
gst_qtdemux_add_stream() to _create_stream(). This ensures the segment
is always initialized when the stream is created.
Otherwise the segment format is left as GST_FORMAT_UNDEFINED in the case
were a track is reparsed and qtdemux_reuse_and_configure_stream() is
called instead of gst_qtdemux_add_stream(). (See
qtdemux_expose_streams() in the non streams-aware case.)
