generate_rtcp can produce empty packets when reduced size RTCP is turned on.
Skip them since it doesn't make sense to push them and they cause errors with
elements that expect RTCP packets to contain data (like srtpenc).
When seeking back to restore the mdat position a flush is pushed
through and it resets downstream segment information. Make sure
that after the flush (that does a soft reset) a segment will
be pushed again
Fixes regressions spotted at
https://ci.gstreamer.net/job/GStreamer-master-validate/2100/
10 FourCCs generated with GST_MAKE_FOURCC() in gstqtmux.c and atoms.c
already exist in fourcc.h. Don't duplicate these and use them directly.
Plus moving 6 to fourcc.h, to centralize them all.
This fixes seeking if the first entries in the samples table are negative. The
binary search would always fail on this as the array would not be sorted if
interpreting the negative numbers as huge positive numbers. This caused us to
always output buffers from the beginning after a seek instead of close to the
seek position.
Also add a case to the comparison function for equality.
Actual code is checking for a NULL terminator and a ';' terminator,
for backward compat, in a chained way that cause all events being rejected.
The proper condition is to reject the events when terminator isn't
in ['\0', ';'] set.
https://bugzilla.gnome.org/show_bug.cgi?id=758151
It would be unusual to have the header segment with an 'edts' atom
indicating gaps at the beginning when handling fragmented streams.
The header usually doesn't contain any timestamping information, this
should come from the playlist/manifest and the segments with media
in those scenarios.
https://bugzilla.gnome.org/show_bug.cgi?id=758171
On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On Windows it
is a setting for the receiver socket. As such we will need it on udpsrc too to
allow filtering out our own multicast packets.
In push-mode it is hard to support qt segments overall but it is
possible to support when the file isn't heavily edited but just contain
a segment to indicate a gap at the beginning. This also allows properly
timestamping data that has negative DTS in push-mode.
It is relevant to support those for 2 scenarios:
1) fragmented streaming
2) HTTP playback of 'regular' mp4
https://bugzilla.gnome.org/show_bug.cgi?id=753484
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
For the MS/VfW codec ids, we want to write DTS timestamps instead
of PTS because that's what everyone else seems to do (and it's also
how it is in AVI). So for those input formats we use the buffer DTS
instead of the PTS. However, if there's no DTS set but only the PTS
then just take the PTS instead of dropping the input buffer. This
is useful especially for I-frame only codecs like JPEG and huffyuv,
but should also be fine as fallback in general.
Fixes regression with input JPEG frames that only have PTS set on them.
https://bugzilla.gnome.org/show_bug.cgi?id=756967
