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
TIME segment implies that stream/running time is being handled by upstream.
So, we shouldn't override it without any clue.
This patch is for fixing seek in DASH streaming.
https://bugzilla.gnome.org/show_bug.cgi?id=774196
qtdemux.c: In function ‘qtdemux_parse_tree’:
qtdemux.c:10139:16: error: ‘color_table_id’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
if (color_table_id != 0) {
^
qtdemux.c:10121:19: note: ‘color_table_id’ was declared here
guint16 color_table_id;
^~~~~~~~~~~~~~
The ProRes guidelines suggest an interleave of 0.5s is common, but
specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
be used per chunk.
It might also make sense to use similar numbers in general.
https://bugzilla.gnome.org/show_bug.cgi?id=773217
Previously we were switching from one chunk to another on every single
buffer. This wastes some space in the headers and, depending on the
software, might depend in more reads (e.g. if the software is reading
multiple samples in one go if they're in the same chunk).
The ProRes guidelines suggest an interleave of 0.5s is common, but
specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
be used per chunk. This will be handled in a follow-up commit.
https://bugzilla.gnome.org/show_bug.cgi?id=773217
It's required for ProRes to work with other software.
It is also in the MP4 standard, but inventing values here seems a bit
tricky for the general case and it does not really give any extra
information.
https://bugzilla.gnome.org/show_bug.cgi?id=769048
Use the number of milliframes per second for integral and drop-frame
framerates, as suggested by the QT file format specification and other
places. We already did that for integral framerates before, but not for
drop-frame framerates. This now keeps precision better.
For all other framerates, check if it's close to a well-known framerate
and use that instead.
https://bugzilla.gnome.org/show_bug.cgi?id=769041
We consider there's a sifnificant difference when it's larger than on second
or than half the duration of the last processed fragment in case the latter is
larger.
https://bugzilla.gnome.org/show_bug.cgi?id=754230
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
And don't just reset everything. This makes sure that we can continue to
handle data in the following scenario:
moov: discont
moof: discont
mdat: continuous
Previously this would fail because the offset would be the accumulated offset
from moov and moof at the mdat position, while the buffer offset might be
something completely different.
As seen in the parent switch for object_type_id, the 4 possible values are
0x40, 0x66, 0x67 and 0x68. Fixing the nested switch to match these values.
Looks like it was a typo making them decimal instead of hexadecimal.
CID 1363328
Without, raw AAC can't be handled and we have some information available in
the decoder that most likely allows us to decode the stream in one way or
another. This is the same code already used by matroskademux for the same
reasons, and ffmpeg/vlc play such files just fine too by guesswork.
This is to handle cases where upstream handles the fragmented streaming in TIME
segments and sends us data with gaps within fragments. This would happen when dealing
with trick-modes.
When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
it must obey the following rules:
* The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
* The buffers containing the first sample after a gap:
* MUST start at the beginning of a sample,
* MUST have the DISCONT flag set,
* MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.
https://bugzilla.gnome.org/show_bug.cgi?id=767354
No variables were added/removed. This was just a good excuse to:
* Comment what most variables are used for (and when)
* Order them in such a way as to show first the common variables used
in all cases, followed by those only used in push-mode
We shouldn't go through segment activation as we will only have a limited
understanding of how the whole stream timeline looks like from the moof. We
only know about the current fragment, while upstream knows about the whole
stream.
This fixes seeking in DASH streams, both for seeks after the current moof and
for seeks into the current moof. The former would fail because the moof ends
and we can't activate any segment, the latter would cause a segment that stops
at the moof end, and no further fragments would be played because we end up
being EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=767071
segment_duration and media_time should be parsed based on version
of elst box. Specification defines that an elst box with version 1
has uint64 and int64 values for segment_duration and media_time,
respectively.
https://bugzilla.gnome.org/show_bug.cgi?id=766301
Properly handle edts segments for push-based operation seeking.
We only support edts that a single segment that has media at the end,
being preceeded by any number of gap segments.
This also allows the qt segment rate to be respected after seeks
https://bugzilla.gnome.org/show_bug.cgi?id=765669
Via the MPEG-4 Part 3 spec we can support the other layers too.
Also correct the samples per frame calculation for MP3 if it's MPEG-2 or
MPEG-2.5.
https://bugzilla.gnome.org/show_bug.cgi?id=765725
timescale/1 is unreliable value for framerate. Due to downstream
element usually use framerate generated by qtdemux, let it be omitted
until the framerate can be reliably calculated.
https://bugzilla.gnome.org/show_bug.cgi?id=764733
When playing a stream that has been protected by DASH CENC, playback
will fail if a seek is performed. Qtdemux produces the error "stream
is protected using cenc, but no cenc protection system information
has been found" and playback stops.
The problem is that gst_qtdemux_reset() gets called as part of the
FLUSH during a seek. This function frees the protection_system_ids
array. When gst_qtdemux_configure_protected_caps() is called after the
seek has completed, the protection_system_ids array is empty and
qtdemux is unable to create the correct output caps for the protected
stream.
This commit changes it to only free the protection_system_ids on
hard resets.
https://bugzilla.gnome.org/show_bug.cgi?id=761787
qtdemux->streams is an array, it will never evaluate to true when comparing
to NULL. Instead we want to check the number of streams to make sure the
stream is available.
https://bugzilla.gnome.org/show_bug.cgi?id=753614
CID 1358389
The PIFF data is stored in a custom UUID box which is parsed and the
crypto_info of the element is updated accordingly. This allows
downstream decryptors to process and decrypt the protected content.
https://bugzilla.gnome.org/show_bug.cgi?id=753614
If we don't find the index of the sample correctly in src_convert function,
we have to unref about the qtdemux before returning value.
So, I have modify it about instead pass qtdemux as a parameter into
src_convert function.
https://bugzilla.gnome.org/show_bug.cgi?id=763973
Currently, get_duration function always return the TRUE even though
it can't be set duration correctly. So, we need to add the else condition
about the fail case. Also, we already set the GST_CLOCK_TIME_NONE
in this function. So I have modify it which is related code in some
function.
https://bugzilla.gnome.org/show_bug.cgi?id=763968
When upstream is running in bytes in push-mode, qtdemux will
convert seeks from time to bytes and send it upstream. Upstream
element will perform a byte seek and send a byte segment to qtdemux
that will convert it to time and push it downstream.
There is, however, the pending_segment variable that stores a new
segment event to be pushed before the next data. When handling seeks
as mentioned above this variable was being ignored and, if it contained
some segment event, it would override the one resulting from the seek.
This would restore a previous segment and would cause the seek segment
to be discarded downstream.
This patch fixes this issue by unrefing any pending segment as the
seek from upstream should contain the latest one that should be
used, as requested by the application.
https://bugzilla.gnome.org/show_bug.cgi?id=763165
In some cases the stream configuration can fail, for instance if the
stream is protected and no decryptor was found. For those situations
the demuxer shouldn't emit any data on the corresponding source pad of
the stream and bail out.
https://bugzilla.gnome.org/show_bug.cgi?id=762516
When the cenc metadata is stored outside of the moof box and the
stream is exposed it is possible that the cenc metadata hasn't been
processed yet while the first buffer is being pushed. When this
happens the buffer can't possibly be decrypted downstream so don't
push it.
https://bugzilla.gnome.org/show_bug.cgi?id=762516
Commit 7873bede31
(https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the
behaviour of qtdemux to call gst_qtdemux_configure_stream() for
every new moof.
When playing a protected stream, gst_qtdemux_configure_stream()
calls gst_qtdemux_configure_protected_caps(). The
gst_qtdemux_configure_protected_caps() function takes the original
media format, puts this in a field called "original-media-type"
and then changes the caps to "application/x-cenc".
The gst_qtdemux_configure_protected_caps() did not handle the case
of being called multiple times, causing it to incorrectly set the
caps. The second call was causing the caps to be set to:
application/x-cenc, original-media-type"application/x-cenc"
This commit makes gst_qtdemux_configure_protected_caps() check that
the caps have already been transformed, so that it only gets
changed once.
https://bugzilla.gnome.org/show_bug.cgi?id=761769
qtdemux calculates framerate using duration and the number of sample.
In case of fragmented mp4 format, however, the number of sample can
be figure out after parsing every moof box. Because qtdemux does not
parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect
framerate calculation.
This patch will triger gst_qtdemux_configure_stream() for every new moof.
Then, framerate will be calculated by using duration and n_samples of the moof.
https://bugzilla.gnome.org/show_bug.cgi?id=760774
Based on document ISO_IEC_14496-12, edit list box can have
segment duration as zero. It does not imply that media_start equals to
media_stop. But, it just indicates a sample which should be presented
at the first. This patch derives segment duration using media_time
and duration of file. And set derived duration to segment-duration.
https://bugzilla.gnome.org/show_bug.cgi?id=760781
In case of push mode, qtdemux expose streams after got moov box.
We can not guarantee that a moov box has sample data such as sample duration
and the number of sample in stbl box for fragmented format case.
So, if a moov has no sample data, streams will not be exposed until get the first moof.
https://bugzilla.gnome.org/show_bug.cgi?id=760779
Prevents downstream from receiving flushes for a seek only in
upstream. Those seeks are only to start reading from the right
offset when skipping or returning to qt atoms.
https://bugzilla.gnome.org/show_bug.cgi?id=758928
Avoid writing a negative number as a large positive
integer in an edit list when the first_ts is smaller
than the first_dts - which can happen when the first
packet received has a PTS but no DTS.
https://bugzilla.gnome.org/show_bug.cgi?id=759615
When working in push-mode, we attempt to push out everything currently
buffered in the adapter.
This has two pitfalls:
* We could stop earlier (the moment we get a non-ok or non-not-linked)
* We return the last combined flow return, which might be completely
different from the previous combined flow return
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.
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
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
If the QtDemuxStream are re-used they may already have caps which used
to be leaked.
Reproduced using the
validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=756561
Negotiation to audio/x-raw,format=S8 was not possible because S8 does
not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
https://bugzilla.gnome.org/show_bug.cgi?id=756387
If seeking targets an empty segment skip it as there is no media
offset to get from it. Instead look for the next one.
This doesn't make seeking in push-mode work if you seek to an
empty segment but at least won't get you to wrong offsets.
https://bugzilla.gnome.org/show_bug.cgi?id=753484
eceb2ccc73 broke segment seeks by always
accumulating segments manually when activating a segment. This is only
needed when handling edit lists, not when activating a segment because of a
seek. Do the accumulation when switching edit list segments instead.
This fixes segment seeks again, while keeping edit lists playback working.
https://bugzilla.gnome.org/show_bug.cgi?id=755471
Commit 7d7e54ce68 added support for
DASH common encryption, however commit
bb336840c0 that went onto master
shortly before the CENC commit caused the calculation of the CENC
aux info offset to be incorrect.
The base_offset was being added if present, but if the base_offset
is relative to the start of the moof, the offset was being added twice.
The correct approach is to calculate the offset from the start of the
moof and use that offset when parsing the CENC aux info.
This commit adds support for ISOBMFF Common Encryption (cenc), as
defined in ISO/IEC 23001-7. It uses a GstProtection event to
pass the contents of PSSH boxes to downstream decryptor elements
and attached GstProtectionMeta to each sample.
https://bugzilla.gnome.org/show_bug.cgi?id=705991
When a new time segment is received upstream is going to restart
with a new atom. Make the neededbytes and todrop variables
reflect that to avoid waiting too much or dropping the
initial bytes that contain the header.
The adapter might have data remaining from the previous segment,
push it all before clearing the adapter and starting a new segment.
It can accumulate data if it had pushed and got not-linked, returning
immediately without processing all the data. Before starting a new
segment this data should be handled.
Avoids accumulating all samples from a fragmented stream that could
lead to a 'index-too-big' error once it goes over 50MB of data. It
could reach that before 2h of playback so it doesn't take that long.
As upstream elements are providing data in time format they should
be the ones that have more information about the full media index
and should be able to seek if possible.
upstream_newsegment isn't really clear on what it means, it is set
to TRUE when the upstream element sends a segment in TIME format, so
rename it to be more clear about it.
It is important to know this because it means that upstream has
a notion of time and qtdemux is likely being driven by an upstream
element that is reading from a higher level abstraction than a file,
such as a DASH, MSS or DLNA element.
In fragmented streaming, multiple moov/moof will be parsed and their
previously stored samples array might leak when new values are parsed.
The parse_trak and callees won't free the previously stored values
before parsing the new ones.
In step-by-step, this is what happens:
1) initial moov is parsed, traks as well, streams are created. The
trak doesn't contain samples because they are in the moof's trun
boxes. n_samples is set to 0 while parsing the trak and the samples
array is still NULL.
2) moofs are parsed, and their trun boxes will increase n_samples and
create/extend the samples array
3) At some point a new moov might be sent (bitrate switching, for example)
and parsing the trak will overwrite n_samples with the values from
this trak. If the n_samples is set to 0 qtdemux will assume that
the samples array is NULL and will leak it when a new one is
created for the subsequent moofs.
This patch makes qtdemux properly free previous sample data before
creating new ones and adds an assert to catch future occurrences of
this issue when the code changes.
Most files don't contain the values for transposing the coordinates
back to the positive quadrant so qtdemux was ignoring the rotation
tag. To be able to properly handle those files qtdemux will also ignore
the transposing values to only detect the rotation using the values
abde from the transformation matrix:
[a b c]
[d e f]
[g h i]
https://bugzilla.gnome.org/show_bug.cgi?id=738681
The media start has nothing to do with the shift we have applied
but with the value of the first PTS. This is defined as:
Dt(0) = 0
Ct(0) = Dt(0) + CTTS(0)
So the media start is always the first CTTS.
https://bugzilla.gnome.org/show_bug.cgi?id=751361
Allows playing edts editted files with proper synchronization of
streams. This patch fixes the regression introduced by
bf95f93c01 that was added to fix
segment seeks handling.
Having the accumulated_base separated from the main segment.base
allows handling both segment seeks and edts editted files.
https://bugzilla.gnome.org/show_bug.cgi?id=751361
Buffers need not to start at running-time 0 so the last_dts needs
to be the value of the first buffer's dts as it is used to compute
the duration of the buffers. If it was left at 0 the first buffer
would have a larger duration when it shouldn't
https://bugzilla.gnome.org/show_bug.cgi?id=751361
Move the multiview caps calculations to the configure_stream()
function, so the rest of the video info is available, and
use the gst_video_multiview_guess_half_aspect() function to
determine if the half-aspect flag should be set on frame-packed
video.
The cslg atom provide information about the DTS shift. This is
needed in recent version of ctts atom where the offset can be
negative. When cslg is missing, we parse the CTTS table as proposed
in the spec to calculate these values.
In this implementation, we only need to know the shift. As GStreamer
cannot transport negative timestamps, we shift the timestamps forward
using that value and adapt the segment to compensate. This patch also
removes bogus offset of ctts_soffset, this offset shall be included
in the edit list.
https://bugzilla.gnome.org/show_bug.cgi?id=751103
We shift DTS forward to avoid negative timestamps which cannot be
represented with version 0 of the CTTS table. To stick with that
version (backward compatibility), the spec recommend using an
edit list entry to move back the presentation time to where it
should be.
https://bugzilla.gnome.org/show_bug.cgi?id=751242
In presence of a CTTS, the segment start/stop must be offset so
the segment start/stop include the PTS. This is needed since the
PTS cannot be negative in this format. This fixes issues where the
running time of the first buffer isn't at the start.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
Remove a custom specialized version of gst_buffer_new_wrapped by
using gst_buffer_new_wrapped_full inside a macro to simplify
parameters and give it a more meaningful name.
It is only used to create temporary buffers to have its data copied.
Adds AC-3 muxing support. It is defined for mp4 and 3gp formats.
One extra feature that was added was the ability to add extension
atoms after set_caps as the AC-3 extension atom needs some data
that has to be extracted from the stream itself and is not
present on caps.
The MPEG-A format provides an extension to the ISO base media
file format to store stereoscopic content encoded with different
codecs like H.264 and MPEG-4:2. The stereo video media information(svmi)
atom declares the presence and storage method for the video.
Stereo video information for MPEG-A can also be supplied through
the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which
is not implemented in this patch.
Also missing is support for stereo video encoded as separate video tracks
for now.
Based on a patch by Sreerenj Balachandran <sreerenj.balachandran@intel.com>
https://bugzilla.gnome.org/show_bug.cgi?id=611157
When performing seek, segment->start is being updated with desired_offset,
but in case of reverse playback segment->start should be 0 and
segment->stop should be updated with desired offset.
https://bugzilla.gnome.org/show_bug.cgi?id=750675
Only update the moov header into the caps if it's the finalised
moov at EOS time. Avoids posting a bogus moov at startup and
repeated updates in robust-recording mode
Implement a robust recording mode, where the output
file is always in a playable state, seeking and rewriting
the moov header at a configurable interval. Rewriting
moov is done using reserved space at the start of
the file, and a ping-pong strategy where the moov
is replaced atomically so it's never invalid.
Track when tags have actually changed, and don't write them into
the moov unless they've changed. Clear any existing tags when
re-writing them, so we can do progressive moov updating in robust
recording mode.
Write placeholder mdat as a free atom plus a 32-bit mdat
with '0' size, which means "rest of the file" in the spec.
Re-write it later to a full 64-bit extended size atom if needed.
Correctly update any edit lists each time the moov is recalculated,
updating existing table entries if they already exist instead of just
adding new ones.
qtdemux creates a samples array and gets the timestamps for buffers by
accumulating their durations. When doing reverse playback of fragments,
accumulating samples will lead to wrong timestamps as the timestamps
should go decreasing from fragment to fragment and the accumulation
will produce wrong results.
In this case, when receiving a discont for fragmented reverse playback,
the previous samples information should be flushed before new data
is processed.
The gst-launch script for example launch line to test qtdemux is
missing a queue before the decodebins, otherwise the gst-launch-1.0
command won't work.
https://bugzilla.gnome.org/show_bug.cgi?id=749054
When doing key uint seek, qtdemux calls gst_qtdemux_adjust_seek
to get proper offset. And then this offset is set to
segment.position and segment.time in gst_qtdemux_perform_seek but
segment.start is not updated.
After that, application sends segment query,
qtdemux sets start and stop to query using gst_segment_to_stream_time. Due
to the wrong value in segment.start, the stop position is smaller than
it should.
https://bugzilla.gnome.org/show_bug.cgi?id=746822
We always write the CTTS in qtmux. Ideally we only want to do that
for streams that need DTS, it should be present on the track information
rather than be decided based on each buffer
As qt uses durations, it doesn't matter, only the difference
between consecutive buffers is important. Also, collectpads
already replaces PTS/DTS with the running times for them.
Instead of checking various state variables around the muxer,
track the current muxing mode in a single 'mux_mode' enum.
Add some implementation notes about the different mux modes
gst_segment_do_seek() does that for us already, and doing it twice
will break non-flushing seeks in interesting ways. Leftover from 1.0
porting.
Also copy over segment offset and applied_rate, just in case.
When not in fast-start or fragmented mode, we need to be able
to rewrite the size of the mdat atom, or else the output just
won't be playable - the mdat placeholder with size == 0 will
cover the rest of the file, including any moov atom we write out.
https://bugzilla.gnome.org/show_bug.cgi?id=708808
Unlike many other seek flags, the KEY_UNIT seek
flag is not copied over into the GstSegment,
since it's only relevant for the seek itself,
so we need to pass it explicitly to the seek
handler here.
https://bugzilla.gnome.org/show_bug.cgi?id=745339
We need different symbol names, because these symbols are also present
in the fragmented plugin ... which will cause conflicts when doing
static linking
Using the sparse streams can make the push-based seeking return
too far in the stream. It also can lead to issues as the
sparse streams will be ignored when restarting playback and,
if the sparse stream is the one that has the earliest sample,
it will confuse qtdemux's offsets as one stream will have
an earlier offset than the demuxer's one which might lead to
early EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=742661
Parse the 'sidx' atom and update the total duration according to the
parser result. The isoff parser code is imported from
gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
function was factored out of the gst_isoff_sidx_parser_add_buffer()
function.
https://bugzilla.gnome.org/show_bug.cgi?id=743578
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
When writing out a trak with an edit list, make sure the
overall file duration is also updated to reflect the
lengthening of the stream.
Add some more debug to qtdemux to warn about streams that
are longer than the file and get truncated.
Handle the transformation matrix cases where there are only simple rotations
(90, 180 or 270 degrees) and use a tag for those cases. This is a common scenario
when recording with mobile devices
https://bugzilla.gnome.org/show_bug.cgi?id=679522
