Sample table based segment event (genereted by qtdemux) could break
presentation timeline. For example, qtdemux should not modify upstream
time format segment (e.g., adaptivedemux use case)
https://bugzilla.gnome.org/show_bug.cgi?id=796480
This field is actually only informatory and the user can potentially
choose something else. EME tests in WebKit testsuite actually doesn't
take it into and force another encryption system to be used, and expects
to be given the occasion to do so.
This basically also reverts 3e063703b3.
Instead of always keeping a safe segment (start=0) event from the beginning,
delay the creation of this event to when we really know the timestamp of the
first sample. This is important to properly start fragmented streams that
we might join in the middle or to play isolated fragment files that might
have an advanced tfdt.
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Fragmented files often use elst.duration=0 which before
ee78825eae was wrongly interpreted as
having no frames.
Since that issue has now been fixed, there is no reason to disable edit
lists in fragmented files. This commit enables them, therefore producing
correct stream time for files containing edit lists.
https://bugzilla.gnome.org/show_bug.cgi?id=793058
Since ca068865c3 the duration of the first
frame is not used for estimating the frame rate.
For this purpose, stream->first_duration was initialized with the
duration of the first frame. In fragmented files, this was previously
done by peeking the first moof, but that can only be done in pull mode.
Fortunately, we don't really need to do that, at least with the current
design: When we are estimating the frame rate we already have the
sample table, regardless of the scheduling mode and whether the file is
fragmented or not, so we can obtain first_duration there much more
reliably.
This fixes frame rate estimation for fragmented files in push mode.
https://bugzilla.gnome.org/show_bug.cgi?id=796384
Whenever got new moov or new stream-start,
demux will try to expose new pad by following rule.
Comparing stream-id in the current moov with previous one, then
* If matched stream-id is found from previous one,
reuse existing pad (most common case)
* Otherwise, expose new pad with new stream-start
* No more used stream will be freed
https://bugzilla.gnome.org/show_bug.cgi?id=684790
Whenever demux got moov, demux will create new stream. Only exception is
duplicated track-id in a moov box. In that case the first stream
will be accepted. This patch is pre-work for rework of moov handling.
https://bugzilla.gnome.org/show_bug.cgi?id=684790
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.
Corrupted files could potentially have multiple cdat/cdt2 atoms in
a sample entry, which is unclear how to handle.
Ignore repeated ones.
CID #1434162
CID #1434159
The value stored in cenc_aux_sample_count wasn't in sync with the
parsing code that followed which checks whether all entries are
valid and present.
Only write the actual sample count when we know for sure.
CID #1427087
The samples table is sorted by DTS, not PTS. As such we can only get the
correct result when using a binary search on it, if we search for the
DTS.
Also if we only ever search for the frame, where the following frame is
the first one with a PTS after the search position, we will generally
stop searching too early if frames are reordered.
In forwards playback this is not really a problem (after the decoder
reordered the frames, clipping is happening), in reverse playback
it means that we can output one or more frames too few as we stop too
early and the decoder would never receive it.
https://bugzilla.gnome.org/show_bug.cgi?id=782118
The smallest possible is 24 (and not 25) bytes.
The last "name" field can according to QTFF specifications not be present
at all. The parser will handle this fine and so will the rest of
the qtdemux code.
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 we saw empty segments, we previously unconditionally pushed a
GAP event downstream regardless of the duration of that empty
segment.
In order to avoid issues with initial negotiation of downstream elements
(which would negotiate to something before receiving any data due to
that initial GAP event), check if there's at least a second of difference
(like we do for other GAP-related checks in qtdemux) before
deciding to push a GAP event downstream.
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.
It is possible that the mdat has more data than what was stored in the
headers file. If we put that to the output the file will have bogus data
at the end and some players will complain.
https://bugzilla.gnome.org/show_bug.cgi?id=784258
qtdemux.c: In function ‘gst_qtdemux_configure_stream’:
qtdemux.c:7764:34: error: suggest parentheses around ‘&&’ within ‘||’ [-Werror=parentheses]
if ((stream->n_samples == 1) && (stream->first_duration == 0)
~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Avoid computing frame rate when a stream contain moof with only one
sample, to avoid an assert. The moof is considered as still picture.
The same is already done for one sample given in the moov.
https://bugzilla.gnome.org/show_bug.cgi?id=782217
... which no longer worked due to unconditionally clearing sample info and
ending up in inconsistent state. Let's tread a bit more carefully and also
allow for the old seek handling that resorts to scanning if no mfra info
is available.
The last entry will most likely get new samples added to it in "robust"
muxing mode, changing the samples_per_chunk and thus making it wrong to
keep the last two entries merged. It will run into an assertion later
when adding a new sample to the chunk.
Thanks to gdiener@cardinalpeak.com for the analysis of the bug and
proposal for a solution.
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
In push mode we process as much as possible in the adapter. When we receive
a DISCONT buffer which we can't match to an actual sample (based on the existing
sample table) and there is still data remaining in the incoming adapter,there is
one of two cases happening:
1) We are doing reverse playback, in which case we should flush out all pending
data
2) We have leftover data from the previous incoming buffer... which we can't do
anything about.
For the second case, make sure we flush out the remaining data so that we can start
parsing again from scratch.
https://bugzilla.gnome.org/show_bug.cgi?id=781319
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.
Make sure offset and neededbytes are properly resetted when all
streams are EOS in push-mode.
Avoids cases when some data might still be pushed by upstream (because
it didn't yet see the resulting GST_FLOW_EOS yet) and qtdemux gets
completely lost.
https://bugzilla.gnome.org/show_bug.cgi?id=781266
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
Before pushing a sample, check if there was a change in the current
stsd entry. This patch also assumes that the first stsd entry is
used as default for the first sample. It might cause an uneeded
caps renegotiation when this isn't the case.
stsd can have multiple format entries, parse them all.
This is required to play DVB DASH profile that uses multiple entries
to identify the different available bitrates/options on dash streams
The stream format-specific data is not stored into QtDemuxStreamStsdEntry
Instead of using the stsd as a base pointer, use the actual stsd
entry as the stsd can have multiple entries. This is rarely used
for file playback but is a possible profile with in DVB DASH specs.
This still doesn't support stsd with multiple entries but makes it
easier to do so.
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.
This is an update of d78d589627
We still exit as early as possible in case of non-ok/non-unlinked combined
flow, but we first make sure that we update the internal position variables.
This ensures that if upstreams "ignores" the flow return (and carries on pushing),
we don't end up processing data with completely bogus variables/positions.
TFDTs with time 0 are being ignored since commit 1fc3d42f. They're
mistaken with the case of not having TFDT, but those two cases
must be distinguished in some way.
This patch passes an extra boolean flag when the TFDT is present.
This is now the condition being evaluated, instead of checking for
0 time.
https://bugzilla.gnome.org/show_bug.cgi?id=780410
If we have multiple tracks with timecodes, or it's not the first track
that has timecodes, or not the first buffer, we already started a chunk
for media data. We now need to "close" that chunk because we wrote data
for the timecode track and a new chunk has to be started for the
original track the next time it has data.
Similar to what was done in adaptivedemux, ignore seek
events we've already handled - such as when they are received
on every srcpad of files with lots of streams.
Otherwise mdatleft will have a value calculated from the initial
mdatsize minus the parts of the stream that we saw, which is not
including all the parts of the stream that might've been skipped.
Take into account the atoms at the end of the 'trak' atom when
recovering it. So that its size (already computed and added in the trak
size) isn't making offsets wrong.
https://bugzilla.gnome.org/show_bug.cgi?id=771478
We parse the next moof in advance of having pushed
all samples from the previous one in some cases, and
we'll still need the crypto info from the previous
fragment so keep around any unused crypto info entries
when adding new ones
qtdemux.c: In function ‘qtdemux_parse_samples’:
qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’ instead [-Werror=int-in-bool-context]
if (stream->samples_per_frame * stream->bytes_per_frame) {
~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
This reverts commit 107902ec51.
This commit intended to ensure that keyframe seeks land at the
start timestamp of a keyframe, rather than in the middle of one,
but they cause trouble on files with sparse streams, or with
JPEG 'cover art' tracks that have only one or a few JPEG samples
with very long durations.
That's still desirable for doing seamless cutting of videos,
but needs a rethink for implementation.
https://bugzilla.gnome.org/show_bug.cgi?id=778690
The n_frames field (frames per second) should follow the nominal frame
rate for drop-frame timecodes.
Also, the trak's timescale (and duration, accordingly) should follow the
STSD entry's timescale and frame duration (fps_n and fps_d accordingly),
not the other way around.
https://bugzilla.gnome.org/show_bug.cgi?id=777832
qtdemux_handle_xmp_taglist() requires a writable taglist,
but qtdemux->tag_list can become non-writable, specifically
after sending global tags (qtdemux.c:958), which adds a
second reference. Ensure the list is made writable before
calling (make_writable will copy the list if necessary).
https://bugzilla.gnome.org/show_bug.cgi?id=766177
These are usually much bigger than icon size and required by
iTunes to be certain fairly large sizes. In qtmux it is also
the IMAGE tags which we write out as 'covr' atoms.
When reset, don't restart request pad numberings, as
request pads can survive across state changes. Only
restart at 0 if all request pads are handed back first.
https://bugzilla.gnome.org/show_bug.cgi?id=777174
'stream-format' and 'alignment' are defined in pad template caps so
there is no need to check them again here. Also remove bitrate parsing from
caps as bitrate in caps doesn't make sense but from tags, which is
actually the case.
https://bugzilla.gnome.org/show_bug.cgi?id=777181
Needed for QuickTime 7 to properly play files.
Also write the clap atom for MOV files always, not only when ProRes is
used as a video codec. It's mandatory for MOV.
https://bugzilla.gnome.org/show_bug.cgi?id=777100
The seqh buffer allocated in qtdemux_parse_svq3_stsd_data() needs to
be freed by the caller after use.
https://bugzilla.gnome.org/show_bug.cgi?id=777157
Signed-off-by: Andre McCurdy <armccurdy@gmail.com>
If a fragmented stream doesn't have a tfdt, don't
reset the output timestamps at each fragment boundary
by erroneously using the default value of 0. Introduced
by commit 69fc48
https://bugzilla.gnome.org/show_bug.cgi?id=754230
When performing a key-unit seek, always snap to the start ts
of the keyframe buffer we landed on so that the keyframe is
entirely within the resulting outgoing segment. That seems
the most sensible result, since the user requested snapping
to the keyframe position.
Segments times and seek requests are stored and handled
in raw 'PTS' time, without the cslg_shift - which only applies
to outgoing samples. Omit the cslg_shift portion when
extracting PTS to compare for internal seek snaps.
If the cslg_shift is included, then keyframe+snap-before seeks
generate a segment start/stop time that already includes the
cslg_shift, and it's then added a 2nd time, causing the
first buffer(s) to have timestamps that are out of segment.
Remove an old check from atom_stsc_add_new_entry() that
extends the last entry in the STSC if the samples per chunk
matches, as the new interleave merging logic requires that
the final entry by updateable. There's already code
below which simply merges the final entry into the previous
one when needed, so rely on that instead.
Fixes asserts like:
ERROR:atoms.c:2940:atom_stsc_update_entry: assertion failed:
(atom_array_index (&stsc->entries, len - 1).first_chunk == first_chunk)
We can't simply assume that the length of the tag value as given
inside the stream is correct but should also check against the amount of
data we have actually available.
https://bugzilla.gnome.org/show_bug.cgi?id=775451
qtdemux.c: In function ‘qtdemux_parse_trak’:
qtdemux.c:10184:38: error: format ‘%lu’ expects argument of type ‘long unsigned int’, but argument 9 has type ‘gint {aka const int}’ [-Werror=format=]
GST_DEBUG_OBJECT (qtdemux, "Found jpeg: len %u, need %lu", len,
^
39f7e52266 was setting the buffer duration
to 0 if is not valid, under the assumption that this is "the last"
buffer and no others are coming next. This is wrong, last_buf is the
previous buffer and not the very last one.
4e3c13c87c was setting DTS to 0 if there
was none. This will set DTS to 0 for all e.g. audio streams, completely
messing up calculations if streams don't start at 0.
https://bugzilla.gnome.org/show_bug.cgi?id=774840
| ../../../git/gst/isomp4/qtdemux.c: In function 'qtdemux_parse_tree':
| ../../../git/gst/isomp4/qtdemux.c:10224:24: error: 'size' may be used uninitialized in this function [-Werror=maybe-uninitialized]
| offset += size;
| ^~
| ../../../git/gst/isomp4/qtdemux.c:10197:25: note: 'size' was declared here
| guint32 size, tag;
| ^~~~
https://bugzilla.gnome.org/show_bug.cgi?id=774747
Always write an edit list for the whole track. In general this is not
necessary except for the case of having a gap or DTS adjustment but
it allows to give the whole track's duration in the usually more
accurate media timescale.
https://bugzilla.gnome.org/show_bug.cgi?id=774403
