Keep global and stream tags separately and parse the udta node
that can be found under the trak atom. The udta will contain
stream specific tags and will be pushed as such
https://bugzilla.gnome.org/show_bug.cgi?id=692473
Tags received via events, when marked as stream tags, will
be stored on that stream's trak atom instead of being stored
in the main tags atom. This allows the resulting file to have
global and stream tags stored.
https://bugzilla.gnome.org/show_bug.cgi?id=692473
Refactor the functions that were bound to the 'moov' atom to
directly pass the desired 'udta' that should receive the tags.
This allows the tags to be written to 'udta' at the 'moov' or
the 'trak' level, creating tags that are for the container or
for a stream only.
https://bugzilla.gnome.org/show_bug.cgi?id=692473
For fragmented streams with extra data at the end of the mdat
qtdemux was not dropping those bytes and would try to use
that extra data as the beginning of a new atom, causing the
stream to fail.
https://bugzilla.gnome.org/show_bug.cgi?id=743407
Actually copy the codec data instead of copying nothing
and then bombing out because there's no data.
Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=741783
When dealing with fragmented files, we will get more accurate duration
information via the mfra and moof atoms.
In order for playback to not stop at the initial duration (from the
moov atom), we need to check and update the various duration variables
when we find more information.
Fixes playback of fragmented files in pull mode
When seeking or finding the previous keyframe, do
comparisons against targets and segments using composition time
to correctly decide which sample times match.
Currently during header parsing, we scan through the entire file
and skip every moof+mdat chunk for fragmented mp4s, which makes
start-up incredibly slow. Instead, just stop at the first moof
chunk when have a moov, and start exposing the streams, so we
can go and start handling the moofs for real.
Empty segments in an edit list have a media_start time of -1,
as they don't actually play any media. Allow for that when
aligning to the reference stream in reverse play.
DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)
https://bugzilla.gnome.org/show_bug.cgi?id=737095
The old default timescale of 1 millisecond produces irrational
numbers for a lot of framerate/audio-packet-duration multiples.
1/1800 is a nicer number, as it tends to produce better fractions
and therefore slightly higher accuracy overall
Change the way the output framerate is calculated
to ignore the first sample (which is sometimes truncated
in my testing) and use the new gst_video_guess_framerate()
function to recognise common standard framerates better.
Remove the code that was sorting the first 20 sample
durations and then ignoring the result.
This makes sense in DASH reverse playback, where the upstream dashdemux
will download DASH segments in reverse order, but push their buffers
forward to qtdemux and mark each segment start as DISCONT. This needs
to be forwarded downstream to the parser/decoder, otherwise it won't work.
https://bugzilla.gnome.org/show_bug.cgi?id=734443
