This duration is equal to the longest duration in all track's tkhd atoms, which
may be comprised of the sum of all edit lists in each track. Empty edit lists
in tracks represent start_time, and the actual media duration is stored in the
mdhd atom.
This change lets the generic demux code derive the longest track duration taken
from mdhd atoms, so the correct duration and start_time combination will be
reported.
Should fix ticket #9775.
Reviewed-by: zhilizhao(赵志立) <quinkblack@foxmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The field is not specific to Opus.
The mp2fixed encoder signals initial_padding and is used
by both the matroska-encoding-delay test as well as
the lavf-mkv tests which necessitated several FATE ref changes.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Matroska generally requires timestamps to be nonnegative, but
there is an exception: Data that corresponds to encoder delay
and is not supposed to be output anyway can have a negative
timestamp. This is achieved by using the CodecDelay header
field: The demuxer has to subtract this value from the raw
(nonnegative) timestamps of the corresponding track.
Therefore the muxer has to add this value first to write
this raw timestamp.
Support for writing CodecDelay has been added in FFmpeg commit
d92b1b1bab and in Libav commit
a1aa37dd0b. The former simply
wrote the header field and did not apply any timestamp offsets,
leading to desynchronisation (if one uses multiple tracks).
The latter applied it at two places, but not at the one where
it actually matters, namely in mkv_write_block(), leading to
the same desynchronisation as with the former commit. It furthermore
used the wrong stream timebase to convert the delay to the
stream's timebase, as the conversion used the timebase from
before avpriv_set_pts_info().
When the latter was merged in 82e4f39883,
it was only done in a deactivated state that still did not
offset the timestamps when muxing due to "assertion failures
and av sync errors". a1aa37dd0b
made it definitely more likely to run into assertion failures
(namely if the relative block timestamp doesn't fit into an int16_t).
Yet all of the above issues have been fixed (in commits
962d631573,
5d3953a5dc and
4ebeab15b0. This commit therefore
enables applying CodecDelay, fixing ticket #7182.
There is just one slight regression from this: If one has input
with encoder delay where the first timestamp is negative, but
the pts of the part of the data that is actually intended to be
output is nonnegative, then the timestamps will currently by default
be shifted to make them nonnegative before they reach the muxer;
the muxer will then ensure that the shifted timestamps are retained.
Before this commit, the muxer did not ensure this; instead the
timestamps that the demuxer will output were shifted and
if the first timestamp of the actually intended output was zero
before shifting, then this unintentional shift just cancels
the shift performed before the packet reached the muxer.
(But notice that this only applies if all the tracks use the same
CodecDelay, or the relative sync between tracks will be impaired.)
This happens in the matroska-opus-remux and matroska-ogg-opus-remux
FATE tests. Future commits will forward the information that
the Matroska muxer has a limited capability to handle negative
timestamps so that the shifting in libavformat can take advantage
of it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is possible for the trailing padding to be zero, namely
e.g. if the AV_PKT_DATA_SKIP_SAMPLES side data is used
for leading padding. Matroska supports this (use a negative
DiscardPadding), but players do not; at least Firefox refuses
to play such a file. So for now only write DiscardPadding
if it is trailing padding and nonzero.
The fate-matroska-ogg-opus-remux was affected by this.
(I wish CodecDelay would not exist and DiscardPadding would
be used to instead trim the codec delay away (with the Block
timestamp corresponding to the time at which the actually
output audio is output).)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
These are the formats we want/need to use when dealing with the Intel
VAAPI decoder for 12bit 4:2:0, 12bit 4:2:2, 10bit 4:4:4 and 12bit 4:4:4
respectively.
As with the already supported Y210 and YUVX (XVUY) formats, they are
based on formats Microsoft picked as their preferred 4:2:2 and 4:4:4
video formats, and Intel ran with it.
P12 and Y212 are simply an extension of 10 bit formats to say 12 bits
will be used, with 4 unused bits instead of 6.
XV30, and XV36, as exotic as they sound, are variants of Y410 and Y412
where the alpha channel is left formally undefined. We prefer these
over the alpha versions because the hardware cannot actually do
anything with the alpha channel and respecting it is just overhead.
Y412/XV46 is a normal looking packed 4 channel format where each
channel is 16bits wide but only the 12msb are used (like P012).
Y410/XV30 packs three 10bit channels in 32bits with 2bits of alpha,
like A/X2RGB10 style formats. This annoying layout forced me to define
the BE version as a bitstream format. It seems like our pixdesc
infrastructure can handle the LE version being byte-defined, but not
when it's reversed. If there's a better way to handle this, please
let me know. Our existing X2 formats all have the 2 bits at the MSB
end, but this format places them at the LSB end and that seems to be
the root of the problem.
As we already have support for VUYA, I figured I should do the small
amount of work to support VUYX as well. That means a little refactoring
to share code.
This is the alphaless version of VUYA that I introduced recently. After
further discussion and noting that the Intel vaapi driver explicitly
lists XYUV as a support format for encoding and decoding 8bit 444
content, we decided to switch our usage and avoid the overhead of
having a declared alpha channel around.
Note that I am not removing VUYA, as this turned out to have another
use, which was to replace the need for v408enc/dec when dealing with
the format.
The vaapi switching will happen in the next change
IEEE-754 differentiates two different kind of NaNs.
Quiet and Signaling ones. They are differentiated by the MSB of the
mantissa.
For whatever reason, actual hardware conversion of half to single always
sets the signaling bit to 1 if the mantissa is != 0, and to 0 if it's 0.
So our code has to follow suite or fate-testing hardware float16 will be
impossible.
Up until now, ff_wmv2_decode_secondary_picture_header() only
set the mb_type array for non I-pictures, so that the decoding
process uses the earlier values of this array; this affects
the output of the wmv8-x8intra FATE-test (which this patch
therefore updates). These earlier values were set when decoding
earlier frames or when the buffer was initially zero-allocated.
A consequence of this is that the output of this test would be
random if ff_find_unused_picture() would select the unused picture
to return at random. Furthermore decoding from a keyframe onwards
depends upon the earlier state of the decoder.
This patch therefore zeroes said array when decoding an I picture.
(It is not claimed that zero is the right value to fill the array with.
I just don't know.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The "AYUV" format is defined by Microsoft as their preferred format for
4:4:4 content, and so it is the format used by Intel VAAPI and QSV.
As Microsoft like to define their byte ordering in little-endian
fashion, the memory order is reversed, and so our pix_fmt, which
follows memory order, has a reversed name (VUYA).
Firstly, the timestamps generated from framerate are inaccurate for
variable framerate mode.
Secondly, the timestamps always start from zero, while pts/dts can
start from nonzero. FLV demuxer rejects such index with message:
"Found invalid index entries, clearing the index".
Necessitated by 6ca43a9675
and 425b309fa4.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This tests the new "-flags2 icc_profiles" option by making sure the
embedded ICC profile gets correctly detected as sRGB.
Signed-off-by: Niklas Haas <git@haasn.dev>
Same issues apply to it as to -shortest.
Changes the results of the following tests:
- matroska-flac-extradata-update
The test reencodes two input FLAC streams into three output FLAC
streams. The last output stream is limited to 8 frames. The current
code results in the first two output streams having 12 frames, after
this commit all three streams have 8 frames and are the same length.
This new result is better, since it is predictable.
- mkv-1242
The test streamcopies one video and one audio stream, video is limited
to 11 frames. The new result shortens the audio stream so that it is
not longer than the video.
The -shortest option (which finishes the output file at the time the
shortest stream ends) is currently implemented by faking the -t option
when an output stream ends. This approach is fragile, since it depends
on the frames/packets being processed in a specific order. E.g. there
are currently some situations in which the output file length will
depend unpredictably on unrelated factors like encoder delay. More
importantly, the present work aiming at splitting various ffmpeg
components into different threads will make this approach completely
unworkable, since the frames/packets will arrive in effectively random
order.
This commit introduces a "sync queue", which is essentially a collection
of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
and are then released for further processing (encoding or muxing) when
it is ensured that the frame in question will not cause its stream to
get ahead of the other streams (the logic is similar to libavformat's
interleaving queue).
These sync queues are then used for encoding and/or muxing when the
-shortest option is specified.
A new option – -shortest_buf_duration – controls the maximum number of
queued packets, to avoid runaway memory usage.
This commit changes the results of the following tests:
- copy-shortest[12]: the last audio frame is now gone. This is
correct, since it actually outlasts the last video frame.
- shortest-sub: the video packets following the last subtitle packet are
now gone. This is also correct.
The amount of padding samples reported by containers take into account the
extended samplerate in HE-AAC.
Fixes ticket #9671.
Signed-off-by: James Almer <jamrial@gmail.com>
wrapped_avframe_decode() uses an AVFrame as dst in av_frame_move_ref()
after having called ff_decode_frame_props() to attach side-date
to this very frame. This leaks all the side-data and metadata
that ff_decode_frame_props() has attached.
This happens in various fate-filter-metadata tests since
6ca43a9675.
These particular leaks (which affect metadata-only)
could be fixed by not adding metadata side-data to AVPackets
in libavdevice if they are also available from the AVFrames.
Yet this would break users that extract the metadata from
AVPackets.
The changes to FATE happen because of the way av_dict_set()
works when it overwrites an already existing entry:
It overwrites the entry to be overwritten with the last entry
and adds the new entry at the end. The end result is that
the first entry of the dict is the second-to-last-entry of
the original dict, the last entry of the dict is the last
entry of the old dict and the first count - 2 entries
of the original dict are at positions 1..count - 2 in their
original order.
Reviewed-by: Timo Rothenpieler <timo@rothenpieler.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This avoids an extra copy of potentially quite big video frames.
Instead of copying the entire frames data into a rawvideo packet it
packs the frame into a wrapped avframe packet and passes it through
as-is.
Unfortunately, wrapped avframes are set up to be video frames, so the
audio frames continue to be copied.
Additionally, this enabled passing through video frames that previously
were impossible to process, like hardware frames or other special
formats that couldn't be packed into a rawvideo packet.
Some samples contain Active Format Descriptors, yet the output
of no test depends upon them, so that they are de-facto untested.
So add a dedicated test for them.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Update the still AVIF parser to only read the primary item. With this
patch, AVIF still images with exif/icc/alpha channel will no longer
fail to parse.
For example, this patch enables parsing of files in:
https://github.com/AOMediaCodec/av1-avif/tree/master/testFiles/Microsoft
Adding two fate tests:
1) demuxing of still image with 1 item - this test will pass regardless
of this patch.
2) demuxing of still image with 2 items - this test will fail without
this patch and will pass with patch applied.
Partially fixes trac ticket #7621
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: James Zern <jzern@google.com>
Up until now, updating extradata was very ad-hoc: The amount of
space reserved for extradata was not recorded when writing the
header; instead the AAC code simply presumed that it was enough.
This commit changes this by recording how much space is available.
This brings with it that the code for writing of and reserving space
for the CodecPrivate and for updating it diverges. They are therefore
split; this allows to put other common tasks like seeking to
right offset as well as writing padding (in case the new extradata did
not fill the whole reserved space) to this common function.
The code for filling up the reserved space is smarter than the code
it replaces; therefore it is no longer necessary to reserve more
than necessary just to be sure that one can add an EBML Void element
(whose minimum size is two) lateron. This is the reason for the change
to the aac-autobsf-adtstoasc test.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is possible by using a dynamic buffer to write them;
said dynamic buffer is (re)used and reset as appropriate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
- Introduce ff_draw_init2, which takes explicit colorspace and range
args
- Use lavu/csp and lavfi/colorspace for conversion, rather than the
lavu/colorspace.h macros
- Use the passed-in colorspace when performing RGB->YUV conversions
The upshot of this is:
- Support for YUV spaces other than BT601
- Better rounding for all conversions
- Particular rounding improvements in >8-bit formats, which previously
used simple left-shifts
- Support for limited-range RGB
- Support for full-range YUV in non-J pixfmts
Due to the rounding improvements, this results in a large number of
minor changes to FATE tests.
Signed-off-by: rcombs <rcombs@rcombs.me>
Fixes CID1396405
MSE and PSNR is slightly improved, and some noticable corruptions disappear as
well.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
The cue_sheet.wv sample contains a cue sheet as APE tags,
yet this is not really covered by fate-wavpack-cuesheet
because the metadata does not affect the output of said test.
So add a proper test for this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Since every DLL can use an individual CRT on Windows, having
an exported function that opens a FILE* won't work if that
FILE* is going to be used from a different DLL (or from user
application code).
Internally within the libraries, the issue can be worked around
by duplicating the function in all libraries (this already happened
implicitly because the function resided in file_open.c) and renaming
the function to ff_fopen_utf8 (so that it doesn't end up exported from
the DLLs) and duplicating it in all libraries that use it.
This makes the avpriv_fopen_utf8 / ff_fopen_utf8 function work in
the exact same way as the existing avpriv_open / ff_open, with the
same setup as introduced in e743e7ae6e.
That mechanism doesn't work for external users, thus deprecate the
existing function.
Signed-off-by: Martin Storsjö <martin@martin.st>
Do this by making this test a transcode test.
Also fix the test requirements and don't add this test to FATE_AFILTER;
instead use a new variable and a new target for flvenc-tests.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Accidentally resurrected in fc49f22c3b
and 7711f19eda,
forgotten in 6ebc71847e and
1a6a088f7c or never needed
(filter-aemphasis).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Each of the intermediately generated lena-*.fits files is only used
for exactly one test; so it could be deleted right after the test.
Switching to a transcode test (which is also more natural) achieves
this. It also adds checksums of the intermediate files to the ref-file.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is mostly straightforward. The major complication is that, as a
result of the 16-bit chunk size limitation, ICC profiles may need to be
split up into multiple chunks.
We also need to make sure to allocate enough extra space in the packet
to fit the ICC profile, so modify both mpegvideo_enc.c and ljpegenc.c to
take into account this extra overhead, failing cleanly if necessary.
Also add a FATE transcode test to ensure that the ICC profile gets
written (and read) correctly. Note that this ICC profile is smaller than
64 kB, so this doesn't test the APP2 chunk re-arranging code at all.
Signed-off-by: Niklas Haas <git@haasn.dev>
We re-use the PNGEncContext.zstream for deflate-related operations.
Other than that, the code is pretty straightforward. Special care needs
to be taken to avoid writing more than 79 characters of the profile
description (the maximum supported).
To write the (dynamically sized) deflate-encoded data, we allocate extra
space in the packet and use that directly as a scratch buffer. Modify
png_write_chunk slightly to allow pre-writing the chunk contents like
this.
Also add a FATE transcode test to ensure that the ICC profile gets
encoded correctly.
Signed-off-by: Niklas Haas <git@haasn.dev>
On empty input the awk script was always successful which caused the
filter-refcmp tests to always succeed.
Also fix the command lines for refcmp_metadata compare function because it
needs auto conversion filters, and update reference of test
filter-refcmp-psnr-rgb because it was missed in
a7fc78c1a6 but was never noticed due to the
original issue...
Signed-off-by: Marton Balint <cus@passwd.hu>
Calculate Spatial Info (SI) and Temporal Info (TI) scores for a video, as defined
in ITU-T P.910: Subjective video quality assessment methods for multimedia
applications.
The range parameters need to be set up before calling
sws_init_context (which selects which fastpaths can be used;
this gets called by sws_getContext); solely passing them via
sws_setColorspaceDetails isn't enough.
This fixes producing full range YUV range output when doing
YUV->YUV conversions between different YUV color spaces.
Signed-off-by: Martin Storsjö <martin@martin.st>
The IMF demuxer does not set the DTS and PTS of packets accurately in all
scenarios. Moreover, audio packets are not trimmed when they exceed the
duration of the underlying resource.
imf-cpl-with-repeat FATE ref file is regenerated.
Addresses https://trac.ffmpeg.org/ticket/9611
The sample mpeg4/mpeg4_sstp_dpcm.m4v existed in the FATE-suite,
but it was surprisingly unused.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This long-existing feature calculates subtitle durations by keeping
it around until the following subtitle is decoded, and then utilizes
the following subtitle's pts as the end point of the previous one.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
Peeking into the muxing queue can improve the estimate of
the lowest timestamp needed for avoid_negative_ts in case
the lowest timestamp is in a packet other than the first packet
to be muxed.
This fixes tickets #4536 and #5784 as well as the output from
the matroska-avoid-negative-ts FATE-test.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
write_packet() has code to shift the packets timestamps
to make them nonnegative or even make them start at ts zero;
this code inspects every packet that is written and if a packet
with negative timestamp (whether this is dts or pts depends upon
another flag; basically: Matroska uses pts, everyone else dts)
is encountered, this is offset to make the timestamp zero.
All further packets will be offset accordingly (with the offset
converted according to the streams' timebases).
This is based around an assumption, namely that the timestamps
are indeed non-decreasing, so that the first packet with negative
timestamps is the first packet with timestamps. This assumption
is often fulfilled given that the default interleavement function
by default interleaves per dts; yet there are scenarios in which
it may not be fulfilled:
a) av_write_frame() instead of av_interleaved_write_frame() is used.
b) The audio_preload option is used.
c) When the timestamps that are made nonnegative/zero are pts
(i.e. with Matroska), because the packet with the smallest dts
is not necessarily the packet with the smallest pts.
d) Possibly with custom interleavement functions.
In these cases the relative sync of the first few packet(s) is offset
relative to the later packets. This contradicts the documentation
("When shifting is enabled, all output timestamps are shifted by
the same amount").
Therefore this commit changes this: As soon as the first packet
with valid timestamps is output, it is checked and recorded whether
the timestamps need to be shifted. Further packets are no longer
checked for needing to be offset; instead they are simply offset.
In the cases above this leads to packets with negative timestamps
(and the appropriate warnings) instead of desync. This will mostly
be fixed in the next commit.
This commit also factors handling the avoid_negative_ts stuff out
of write_packet() in order to be able to return immediately.
Tickets #4536 and #5784 as well as the matroska-avoid-negative-ts-test
are examples of c); as has been said, some timestamps are now negative,
yet the ref file update does not show it because ffmpeg.c sanitizes
the timestamps (-copyts disables it; ffprobe and mkvinfo also show
the original timestamps).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This tests the issue from tickets #4536, #5784;
the output of this test is currently broken.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Tests the parsing and writing of AVDOVIDecoderConfigurationRecord,
when it is present as a Dolby Vision configuration block addition mapping.
Signed-off-by: quietvoid <tcChlisop0@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Up until now, the WebM variant of WebVTT subtitles has been handled
specially: It had its own function to write it, because the data
had to be reformatted before writing. But given that other codecs
also need reformatting, this is no good reason to also duplicate the
generic stuff for writing Block(Group)s.
This commit therefore uses an ordinary reformatting function for
this task; writing WebVTT subtitles now uses the generic code
and therefore automatically uses the least amount of bytes
for its BlockGroup length fields whereas the earlier code used
an overestimation for the length of the Duration element.
This is the reason for the changes to the webm-webvtt-remux FATE-test.
(This commit does not implement support for Matroska's way of muxing
WebVTT; it also does not add checks to ensure that WebM-style subtitles
don't get muxed in Matroska. But the function for reformatting gets a
webm prefix to indicate that this is for WebM.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This commit uses the new EbmlWriter API to write the length fields
of the BlockGroup and its descendants that are themselves Master
elements (namely BlockAdditions and BlockMore) on the least amount of
bytes.
This fixes regressions introduced when the special code for writing
general subtitles was removed. Accordingly, the binsub-mksenc and
matroska-zero-length-block FATE-tests have now been reverted back
to their old state again; the advantages of this approach are evident
with the matroska-vp8-alpha-remux test which up until now wrote
all the length fields of all BlockGroups, BlockAdditions and BlockMore
on eight bytes.
Using the EbmlWriter API also allowed to improve locality in
mkv_write_block(): E.g. both DiscardPadding as well as the
BlockAdditional side-data are now directly used to add elements
to the writer whereas the earlier code had to first check
for whether a BlockGroup should be used and then check again
(after the place where a BlockGroup would be opened if one were
used) for whether there is DiscardPadding or BlockAdditional
side-data to write.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Once upon a time, mkv_write_block() only wrote a (Simple)Block,
not a BlockGroup which is needed for subtitles to convey
the duration. But with the introduction of support for writing
BlockAdditions and DiscardPadding (both of which require a BlockGroup),
mkv_write_block() can also open and close a BlockGroup of its own. This
naturally led to some code duplication which is removed in this commit.
This new code leads to one regression: It always uses eight bytes for
the BlockGroup's length field, whereas the earlier code usually used the
lowest amount of bytes needed. This will be fixed in a future commit.
This temporary regression is also the reason for changes to the
binsub-mksenc and matroska-zero-length-block fate tests.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also check the (user-provided) tags for being overlong; the earlier
code had an implicit unchecked size_t->int conversion.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
To trigger this bug, use `paletteuse=dither=bayer:bayer_scale=0`; you will see
that adjacent pixel lines will use the same dither pattern, instead of being
shifted from each other by 32 units (0x20).
One way to demostrate the bug is:
$ convert -size 64x256 gradient:black-white -rotate 270 grad.png
$ echo 'P2 2 1 255 0 255' > bw.pnm
$ ffmpeg -i grad.png -filter_complex 'movie=bw.pnm,scale=256x1[bw]; [0:v][bw]paletteuse=dither=bayer:bayer_scale=0' gradbw.png
Previously: https://www.rm.cloudns.org/img/uploaded/0bd152c11b9cd99e5945115534b1bdde.png
Now: https://www.rm.cloudns.org/img/uploaded/89caaa5e36c38bc2c01755b30811f969.png
This was caused by passing inconsistent color vs (a,r,g,b) parameters to
color_get(), and NBITS being 5 meaning actually hitting the same cache node
does happen in this case, but ONLY if bayer_scale is zero.
The fix is passing the correct color value to color_get().
Also added a previous-failing FATE test; image comparison of the first frame:
Previously: https://www.rm.cloudns.org/img/uploaded/d0ff9db8d8a7d8a3b8b88bbe92bf5fed.png
Now: https://www.rm.cloudns.org/img/uploaded/a72389707e719b5cd1c58916a9e79ca8.png
(on this less synthetic test image, the bug basically causes noise from cache
hits vs misses)
Tested: FATE passes, which exercises this filter but at the default bayer_scale.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
This is similar to the faststart option of the mov muxer, yet
in contrast to it it works together with reserve_index_space
(the equivalent to reserved_moov_size): If the reserved space
does not suffice, the data is shifted; if not, the Cues are
written at the front without shifting the data.
Several tests that cover (not only) this have been added.
Implements #7017.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It returns a pointer inside the fifo's buffer, which cannot be safely
used without accessing AVFifoBuffer internals. It is easier and safer to
use av_fifo_generic_peek_at().
mvhd and tkhd present the post-editlist duration, while mdhd should
have the pre-editlist duration. Regression since c2424b1f3.
Signed-off-by: Martin Storsjö <martin@martin.st>
- No longer mixes u8 and u16 component accesses (this was UB)
- De-duplicated 8->16 conversion
- De-duplicated component -> plane+offset conversion
- De-duplicated planar + packed RGB
- No longer calls ff_fill_rgba_map
- Removed redundant comp_mask data member
- RGB0 and related formats no longer write an alpha value to the 0 byte
- Non-planar YA formats now work correctly
- High-bit-depth semi-planar YUV now works correctly
And expose the parsed values as frame side data. Update FATE results to
match.
It's worth documenting that this relies on the dovi configuration record
being present on the first AVPacket fed to the decoder, which in
practice is the case if if the API user has called something like
av_format_inject_global_side_data, which is unfortunately not the
default.
This commit is not the time and place to change that behavior, though.
Signed-off-by: Niklas Haas <git@haasn.dev>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
To avoid the ref for this growing to a very large size when attaching
the parsed RPU side data. Since this sample does not have any dynamic
metadata, two frames will serve just as well as 100.
Signed-off-by: Niklas Haas <git@haasn.dev>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Adds support for concat demuxer to copy the side data information
from the input file to the resulting file. It will behave like the
metadata copy, where the metadata of the first file is kept in the
the output file.
Extract the current code that already performs the stream side_data
copy into a separate method and reuse the method in the concat demuxer.
Signed-off-by: Gerard Sole <g.sole.ca@gmail.com>
The mpeg4 encoder is slice-threaded and its output depends upon
the number of threads used. Therefore all tests of this encoder
use a hardcoded number of threads (ENC_OPTS in fate-run.sh contains
"-threads 1"; only the vsynth%-mpeg4-thread tests override this
for the mpeg4 encoder, but they also use a hardcoded value to
be consistent across different systems); only the new shortest
and copy-shortest[12] (implicitly due to the sample used) tests
don't and this leads to FATE-failures.
Fix this by explicitly setting the thread count.
Also switch the shortest test to framecrc, because hashing side data
is itchy even though the side data used here (AV_PKT_DATA_QUALITY_STATS)
has a defined endianness.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Currently, the code doing this is spread over several places and may
behave in unexpected ways. E.g. automatic 'default' marking is only done
for streams fed by complex filtergraphs. It is also applied in the order
in which the output streams are initialized, which is effectively
random.
Move processing the dispositions at the end of open_output_file(), when
we already have all the necessary information.
Apply the automatic default marking only if no explicit -disposition
options were supplied by the user, and apply it to the first stream of
each type (excluding attached pics) when there is more than one stream
of that type and no default markings were copied from the input streams.
Explicitly document the new behavior.
Changes the results of some tests, where the output file gets a default
disposition, while it previously did not.
Also covers muxing and demuxing of nonstandard FLAC channel layouts
and the multi-dim-quant option of the FLAC encoder
(all of which was hitherto uncovered).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Provides coverage for the muxer.
(Thanks to tresh for modifying the whitespace commit hook
to allow to push this ref file with tabs.)
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It uses the test-lrc.lrc sample which was added years ago, but never
used until now.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This information is coded in a standard MP4 KindBox and utilizes the
scheme and values as per the DASH role scheme defined in MPEG-DASH.
Other schemes are technically allowed, but where multiple schemes
define the same concepts, the DASH scheme should be utilized.
Such flagging is additionally utilized by the DASH-IF CMAF ingest
specification, enabling an encoder to inform the following component
of the roles of the incoming media streams.
A test is added for this functionality in a similar manner to the
matroska test.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
They already uncovered an uninitialized-value bug in the ATRAC3 code
in the demuxer; and provide coverage for ID3v2.3.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The new format (given in big/little endian forms) matches the
existing X2RGB10 format, except with B and R channels switched.
AV_PIX_FMT_X2BGR10 data often is created by OpenGL programs
whose buffers use the GL_RGB10 internal format.
Signed-off-by: Manuel Stoeckl <code@mstoeckl.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This resolves a problem where conversions from YUV to X2RGB10LE
would produce color values a factor 4 too small, because an 8-bit
value was placed in a 10-bit channel.
Signed-off-by: Manuel Stoeckl <code@mstoeckl.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When a color indexing transform with 16 or fewer colors is used,
WebP uses "pixel packing", i.e. storing several pixels in one byte,
which virtually reduces the width of the image (see WebPContext's
reduced_width field). This reduced_width should always be used when
reading and applying subsequent transforms.
Updated patch with added fate test.
The source image dual_transform.webp can be downloaded by cloning
https://chromium.googlesource.com/webm/libwebp-test-data/
Fixes: 9368
Signed-off-by: James Zern <jzern@google.com>
This muxer was untested up until now; had it been tested, it would
have been obvious that it has been broken for years.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes trac issue #7473.
Removes encoder delay (skip samples) and writes remaining frame samples after EOF to get correct sample count.
Output is now accurate vs players that use Microsoft's codecs (Windows Media Format Runtime).
Tested vs encode>decode WMAv2 with MS's codecs and most sample rate/bit rate/channel/mode combinations in ASF/XWMA.
WMAv1 appears to use the same delay, from FFmpeg samples.
Signed-off-by: bnnm <bananaman255@gmail.com>
subtitles.mak's fate-sub tests utilize a more strict comparator
("rawdiff"), which causes the tests fail in case of white space
differences, such as CRLF vs LF. This in turn causes these
ffprobe-using TTML-in-MP4 tests to fail on non-LF systems such as
Windows or wine.
