Use g_date_time seconds manipulation to allow to cover the quicktime
spec for creation_time. It uses seconds since 1904.
Both paths could be done using the generic approach of seconds since
1904 with GDateTime handling, but the first path using seconds from
1970 should be more commonly found and avoids a few objects creation and
ref/unref, so keep it there for performance.
Additionally, the code for handling seconds since 1970 changed from >
to >= because having 0 seconds since 1970 is also a valid case for that
path to handle.
https://bugzilla.gnome.org/show_bug.cgi?id=707975
Check for GST_SEEK_TYPE_NONE for stop poistion and only update
the stop time if it is requested. Otherwise just maintain whatever
was stored at the segment
https://bugzilla.gnome.org/show_bug.cgi?id=707530
When the segment has a defined stop position, qtdemux should check
when streams reach this position and mark those as EOS. When all
streams are EOS it will return GST_FLOW_EOS to upstream to allow
the pipeline to finish instead of continuously consume buffers
from upstream that are not useful for the segment.
https://bugzilla.gnome.org/show_bug.cgi?id=707530
When handling seeks in push mode, qtdemux converts the seek to bytes
and pushes upstream. It needs to keep track of the seek and the
subsequent segment to be able to map them back to the requested
seek time and properly preserve the segment stop of the seek.
This is done by using the start offset in bytes of the seek,
that should be the same of the segment from upstream. And this
is also backwards compatible with what qtdemux already was using.
https://bugzilla.gnome.org/show_bug.cgi?id=707530
Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
structure for fragmented MP4 called "avc3". The principal difference
between AVC1 and AVC3 is the location of the codec initialisation
data (e.g. SPS, PPS). In AVC1 this data is placed in the initial
MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data
goes in the first sample of every fragment (i.e. the first sample in
each mdat box). The principal reason for avc3 is to make it easier
for client implementations, because it removes the requirement to
insert the SPS+PPS in to the decoder pipeline every time there is a
representation change.
This commit adds support for the "avc3" atom, which is almost identical
to the "avc1" atom, except it does not contain any SPS or PPS data.
https://bugzilla.gnome.org/show_bug.cgi?id=702004
This should make decoders able to precisely push buffers until the stop
time in case they need the next keyframe to do it.
Also, according to gst_segment_clip, it should only push a buffer that
the starting ts is strictly smaller than the segment stop, so we change
the min < comparison for <=
Use the same seqnum on messages and events for derived events.
Fixed for flushes / stream-start / segment after a seek, and segment
after a segment.
Fixes#676242
cca2f555d1 introduces a regression, where the demux segment is not
reset on flush stop, so the next upstream segment event will calculate
an invalid base time on the new segment to be sent downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=704255
This can only reliably work if demuxers have a
separate streaming thread per srcpad. This should be
done in a demuxer base class, which integrates parts
of multiqueue
https://bugzilla.gnome.org/show_bug.cgi?id=701856
bug #700505
Following a representation change that causes a resolution change,
the video decoder fails to decode correctly. Dashdemux detects the
representation change and pushes a new caps event and an
initialization segment (a new moov atom) to the downstream qtdemux,
but it doesn't handle this new moov yet, it will only parse the
first one it receives.
This commit changes qtdemux to accept a new moov in a dash bitstream
switching scenario.
In case qtdemux is handling a mss stream, do not mark the stream to wait
for EOS after a segment. Even if it seems to be the last one according to
the current streams information.
MSS handling is different here because there is another demuxer driving
the pipeline
The samplerate field in the STSD atom is not right for some ALAC files
(usually when audio is 96kHz/24bits), so the audio caps must be
extracted from the codec data.
https://bugzilla.gnome.org/show_bug.cgi?id=700382
Whenever the demuxer has a new caps on a stream, it should set the
new_caps variable to true and a new caps event will be pushed before
the next buffer
