This fixes a bug where in some files mehd.fragment_duration is one unit
less than the actual duration of the fragmented movie, as explained below:
mehd.fragment_duration is computed by scaling the end timestamp of
the last frame of the movie in (in nanoseconds) by the movie timescale.
In some situations, the end timestamp is innacurate due to lossy conversion to
fixed point required by GstBuffer upstream.
Take for instance a movie with 3 frames at exactly 3 fps.
$ gst-launch-1.0 -v videotestsrc num-buffers=3 \
! video/x-raw, framerate="(fraction)3/1" \
! x264enc \
! fakesink silent=false
dts: 999:59:59.333333334, pts: 1000:00:00.000000000, duration: 0:00:00.333333333
dts: 999:59:59.666666667, pts: 1000:00:00.666666666, duration: 0:00:00.333333334
dts: 1000:00:00.000000000, pts: 1000:00:00.333333333, duration: 0:00:00.333333333
The end timestamp is calculated by qtmux in this way:
end timestamp = last frame DTS + last frame DUR - first frame DTS =
= 1000:00:00.000000000 + 0:00:00.333333333 - 999:59:59.333333334 =
= 0:00:00.999999999
qtmux needs to round this timestamp to the declared movie timescale, which can
ameliorate this distortion, but it's important that round-neareast is used;
otherwise it would backfire badly.
Take for example a movie with a timescale of 30 units/s.
0.999999999 s * 30 units/s = 29.999999970 units
A round-floor (as it was done before this patch) would set fragment_duration to
29 units, amplifying the original distorsion from 1 nanosecond up to 33
milliseconds less than the correct value. The greatest distortion would occur
in the case where timescale = framerate, where an entire frame duration would
be subtracted.
Also, rounding is added to tkhd duration computation too, which
potentially has the same problem.
https://bugzilla.gnome.org/show_bug.cgi?id=793959
For 59.94 FPS, it's common to set 60000 as timescale. For that
timescale, if the audio is late by as little as 0:00:00.000016666
(definitely less than one audio sample), lateness gets rounded to 1.
Added a safeguard that allows lateness up to 1 sample with the specific
trak's timescale, to make sure that values less than e.g. one audio
sample won't break the prefill mode. What will happen in this case is
that the audio will get squeezed back to the video's timestamp, which in
practice means that the audio will be 0.000016666 seconds early (with
the patch).
https://bugzilla.gnome.org/show_bug.cgi?id=797133
Supports CEA 608 and CEA 708 CC streams
Also supports usage in "Robust Prefill" mode if the incoming caption
stream is constant (i.e. there is one incoming CC buffer for each
video frame).
https://bugzilla.gnome.org/show_bug.cgi?id=606643
gst_qt_mux_can_renegotiate () gets called everywhere following
that pattern:
return gst_qt_mux_can_renegotiate (ref(self));
This means the reference must be released both in the success
and failure cases, it was only done in the success case.
If codec_data is changed, the stream is no longer valid.
Rather than keeping running when refusing new caps,
this patch send a warning to the bus.
Also fix up splitmuxsink to ignore this warning while changing caps.
https://bugzilla.gnome.org/show_bug.cgi?id=790000
It generally makes not much sense to configure it for all pads/traks at
once as this value is usually different for each of them. As such, add a
new property on the pads in addition to the existing property on the
whole muxer.
https://bugzilla.gnome.org/show_bug.cgi?id=792649
If a reserved-max-duration is set, we should always track
and update the reserved-duration-remaining estimate, even
if we're not sending periodic moov updates downstream for
full robust muxing.
Timecode trak is only supported for mov right now, not for mp4. That
code would otherwise create an invalid trak if the muxed video contained
timecode metadata.
https://bugzilla.gnome.org/show_bug.cgi?id=782684
We only accept new caps if they are basically the same. We don't want to
reset anything as if the caps are new, otherwise various state could get
out of sync with the current run.
We have some padding added after the initial moov, so a bigger updated
moov can be handled to some degree and is expected. Previously we just
ignored the padding and errored out in cases when the padding would've
just been enough.
This sets up a moov with the correct sample positions beforehand and
only works with constant framerate, I-frame only streams.
Currently only support for ProRes and raw audio is implemented but
adding new codecs is just a matter of defining appropriate maximum frame
sizes.
https://bugzilla.gnome.org/show_bug.cgi?id=781447
When muxing raw audio, we have no way of storing timestamps but are just
storing a continuous stream of audio samples. If the difference between
the expected and the real timestamp becomes to big, we should error out
instead of silently creating files with wrong A/V sync.
https://bugzilla.gnome.org/show_bug.cgi?id=780679
They should have ideally the same timescale of the video track, which we
can't guarantee here as in theory timecode configuration and video
framerate could be different. However we should set a correct timescale
based on the framerate given in the timecode configuration, and not just
use the framerate numerator.
buf is the current pad->last_buf value. If ever it gets copied/unreffed,
we need to make sure to write back the new pointer to the last_buf
variable.
Fixes using wrong pointer values in the case of decrasing DTS value
last_buf is the one we're going to write next, not buf. As such we
should check timestamps against that one if there is one to select the
earliest pad.
Also remember the currently selected pad in the very beginning when
storing the first last_buf.
This both solves some edge cases where not the correct next pad was
selected corresponding to the target interleave.
