These fields are ad-hoc and will be deprecated. Use the recently-added
AV_CODEC_FLAG_COPY_OPAQUE to pass arbitrary user data from packets to
frames.
Changes the result of the flcl1905 test, which uses ffprobe to decode
wmav2 with multiple frames per packet. Such packets are handled
internally by calling the decoder's decode callback multiple times,
offsetting the internal packet's data pointer and decreasing its size
after each call. The output pkt_size value before this commit is then
the remaining internal packet size at the time of each internal decode
call.
After this commit, output pkt_size is simply the size of the full packet
submitted by the caller to the decoder. This is more correct, since
internal packets are never seen by the caller and should have no
observable outside effects.
ISOBMFF (14496-12) made this field ('channelcount') in the
AudioSampleEntry structure non-template¹ somewhere before the
release of the 2022 edition. As for ETSI TS 126 244 AKA 3GPP
file format (V16.1.0, 2020-10), it does not seem contain any
references limiting the channelcount entry in AudioSampleEntry
or in its own definition of EVSSampleEntry.
fate-mov-mp4-chapters test had to be adjusted as it output a
mono vorbis stream, which would now be properly marked as such
in the container.
1: As per 14496-12:
Fields shown as “template” in the box descriptions are fields
which are coded with a default value unless a derived
specification defines their use and permits writers to use
other values than the default.
Splits the currently handled subtitle at random access point
packets that can be configured to follow a specific output stream.
Currently only subtitle streams which are directly mapped into the
same output in which the heartbeat stream resides are affected.
This way the subtitle - which is known to be shown at this time
can be split and passed to muxer before its full duration is
yet known. This is also a drawback, as this essentially outputs
multiple subtitles from a single input subtitle that continues
over multiple random access points. Thus this feature should not
be utilized in cases where subtitle output latency does not matter.
Co-authored-by: Andrzej Nadachowski <andrzej.nadachowski@24i.com>
Co-authored-by: Bernard Boulay <bernard.boulay@24i.com>
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
From x86inc:
> On AMD cpus <=K10, an ordinary ret is slow if it immediately follows either
> a branch or a branch target. So switch to a 2-byte form of ret in that case.
> We can automatically detect "follows a branch", but not a branch target.
> (SSSE3 is a sufficient condition to know that your cpu doesn't have this problem.)
x86inc can automatically determine whether to use REP_RET rather than
REP in most of these cases, so impact is minimal. Additionally, a few
REP_RETs were used unnecessary, despite the return being nowhere near a
branch.
The only CPUs affected were AMD K10s, made between 2007 and 2011, 16
years ago and 12 years ago, respectively.
In the future, everyone involved with x86inc should consider dropping
REP_RETs altogether.
The cHRM chunk is descriptive. That is, it describes the primaries that should
be used to interpret the pixel data in the PNG file. This is notably different
from Mastering Display Metadata, which describes which subset of the presented
gamut is relevant. MDM describes a gamut and says colors outside the gamut are
not required to be preserved, but it does not actually describe the gamut that
the pixel data from the frame resides in. Thus, to decode a cHRM chunk present
in a PNG file to Mastering Display Metadata is incorrect.
This commit changes this behavior so the cHRM chunk, if present, is decoded to
color metadata. For example, if the cHRM chunk describes BT.709 primaries, the
resulting AVFrame will be tagged with AVCOL_PRI_BT709, as a description of its
pixel data. To do this, it utilizes libavutil/csp.h, which exposes a funcction
av_csp_primaries_id_from_desc, to detect which enum value accurately describes
the white point and primaries represented by the cHRM chunk.
This commit also changes pngenc.c to utilize the libavuitl/csp.h API, since it
previously duplicated code contained in that API. Instead, taking advantage of
the API that exists makes more sense. pngenc.c does properly utilize the color
tags rather than incorrectly using MDM, so that required no change.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
segment_time and segment_times are defined as duration specifications, not
timestamps, so calculation of segment duration must account for initial
timestamp. Fixed.
FATE ref for segment-mp4-to-ts changed on account of avoiding premature
segment cut at the end of the first segment.
Defined by H.274, this SEI message is utilized by iPhones to save
the nominal ambient viewing environment for the display of recorded
HDR content. The contents of the message are exposed to API users
as AVFrame side data containing AVAmbientViewingEnvironment.
As the DV RPU test sample is from an iPhone and includes Ambient
Viewing Environment SEI messages, its test result gets updated.
Parsing should probably be enabled for all codecs, at least for headers,
but e.g. the AAC parser produces 1-byte packets of zero padding with it,
so I'm just enabling it for EAC3 for the moment.
Current code may, depending on the muxer, decide to use VSYNC_VFR tagged
with the specified framerate, without actually performing framerate
conversion. This is clearly wrong and against the documentation, which
states unambiguously that -r should produce CFR output for video
encoding.
FATE test changes:
* nuv-rtjpeg: replace -r with '-enc_time_base -1', which keeps the
original timebase. Output frames are now produced with proper
durations.
* filter-mpdecimate: just drop the -r option, it is unnecessary
* filter-fps-r: remove, this test makes no sense and actually
produces broken VFR output (with incorrect frame durations).
Commit 18f24527eb accidentally made side data only packets be handled like a
flush request. Fix this regression by effectively ignoring them as was the
original intention.
Signed-off-by: James Almer <jamrial@gmail.com>
Currently, in case of equality on the first color channel, the order of
the ref colors is defined by the hashing function. This commit makes the
sorting deterministic and improve the hierarchical ordering.
Some encoders, like flac, can send side data only packets at the end.
Eventually, said extradata update should ideally be used to update the header
when writting to seekable output, but for now, ignore them.
Should fix the undefined behavior of passing NULL to memcpy().
Signed-off-by: James Almer <jamrial@gmail.com>
PFM (aka Portable FloatMap) encodes its scanlines from bottom-to-top,
not from top-to-bottom, unlike other NetPBM formats. Without this
patch, FFmpeg ignores this exception and decodes/encodes PFM images
mirrored vertically from their proper orientation.
For reference, see the NetPBM tool pfmtopam, which encodes a .pam
from a .pfm, using the correct orientation (and which FFmpeg reads
correctly). Also compare ffplay to magick display, which shows the
correct orientation as well.
See: http://www.pauldebevec.com/Research/HDR/PFM/ and see:
https://netpbm.sourceforge.net/doc/pfm.html for descriptions of this
image format.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: James Almer <jamrial@gmail.com>
This filter, when used in the "pad" mode, currently makes the
distinction between limited and full range solely by testing for YUVJ
pixel formats at link setup time. This is deprecated and should be
improved to perform the detection based on the per-frame metadata.
In order to make this distinction based on color range metadata, which
is only known at the time of filtering frames, for simplicity, we simply
allocate two copies of the "black" frame - one for limited range and the
other for full range metadata. This could be done more dynamically (e.g.
as-needed or simply by blitting the appropriate pixel value directly),
but this change is relatively simple and preserves the structure of the
existing code.
This commit actually fixes a bug in FATE - the new output is correct for
the first time. The previous md5 ref was of a frame that incorrectly
combined full-range pixel data with limited-range black fields. The
corresponding result has been updated.
Signed-off-by: Niklas Haas <git@haasn.dev>
The idea behind last_pkt_props was to store the properties of the last packet
fed to the decoder. Any sort of queueing required by CODEC_CAP_DELAY decoders
that consume several packets before they start outputting frames should be done
by the decoders in question. An example of this is libdav1d.
This is required for the following commits that will fix last_pkt_props in
frame threading scenarios, as well as maintain its contents during flush.
This revers commit 022a12b306.
Signed-off-by: James Almer <jamrial@gmail.com>
The Encoding field (and the \fe tag) allows to limit font selection to
only those fonts declaring support for the specified codepage in their
OS/2's table "Code Page Character Range" field.
Particularly, Encoding=0 means only font's declaring support for "ANSI",
or rather "Latin (Western European)", are allowed to be selected.
Specifying Encoding=1 allows all fonts to be considered.
We do not want to limit font selection, so specify Encoding=1.
NB: at the time of writing libass only partially supports this field,
thus hiding the issue in any libass-based renderer. A VSFilter-based
DirectShow filter or XySubFilter will reveal the issue when a font not
declaring support for latin characters is specified in a style.
Colour values used in ASS files without a "YCbCr Matrix" header set to
"None" are subject to colour mangling, due to how ASS was historically
conceived. A more in-depth description can be found in the documetation
inside libass' public ass_types.h header. The important part is, if this
header is not set to "None", the final output colours can deviate from
the literal value specified in the file. When converting from non-ASS
formats we do not want any colour shift to happen, so let's set the
appropiate header.
NB: ffmpeg's subtitle filter, does not follow libass' documentation
regarding colour mangling, thus hiding the bug. Anything based on
VSFilter, XySubFilter or e.g. mpv do and might show the issue.
(Of course native ASS subs, which _do_ rely on colour mangling won't
work properly with the subtitle filter, but this can be fixed another
time)
It is valid for HEVC; in fact, the ATSC-HEVC spec [1] simply
refers to the relevant H.264 spec.
It is also trivial to implement now: Just move applying AFD
to ff_h2645_sei_to_frame() and stop ignoring AFD when parsing
a HEVC SEI containing it.
A FATE-test for this has been added.
[1]: https://www.atsc.org/atsc-documents/a3412017-video-hevc/
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The encoder is sensitive to changes in precision, and its test target
was a compromise. It was already close to failing on x87 FPUs.
ff_mdct_init used double precision entirely from the scale to computing
the MDCT exp tables. av_tx_init uses single-precision for the scale,
with a small input change which was enough to tip the test into failing on
x87 FPUs.
Increase the fuzz factor in line with other AAC encoder tests to fix.
floating point uses a slightly different predictor technique describe here
http://chriscox.org/TIFFTN3d1.pdf
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This commit enabled assembly code with intel AVX512 VNNI and added unit test for sobel filter
sobel_c: 4537
sobel_avx512icl 2136
Signed-off-by: bwang30 <bin.wang@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
The encoder is fixed point, and uses an MDCT only for analysis. Due
to the slightly different rounding, the encoder makes a different
decision, so the tests have to be adjusted as well.
This patch replaces the transform used in AAC with lavu/tx and removes
the limitation on only being able to decode 960-sample files
with the float decoder.
This commit also removes a whole bunch of unnecessary and slow
lifting steps the decoder did to compensate for the poor accuracy
of the old integer transformation code.
Overall float decoder speedup on Zen 3 for 64kbps: 32%
Fixes ticket #128.
The SVQ1 interframe mean VLC symbols -128 and 128 are incorrectly swapped
in our SVQ1 implementation, resulting in visible artifacts for some videos.
This patch unswaps the order of these two symbols.
The most noticable example of the artiacts caused by this error can be observed in
https://trac.ffmpeg.org/attachment/ticket/128/svq1_set.7z '352_288_k_50.mov'.
The artifacts are not observed when using the reference decoder
(QuickTime 7.7.9 x86 binary).
As a result of this patch, the reference data for the fate-svq1 test
($SAMPLES/svq1/marymary-shackles.mov) must be modified. For this file, our
decoder output is now bitwise identical to the reference decoder. I have
tested patch with various other samples and they are all now bitwise identical.
This enables overriding the rotation as well as horizontal/vertical
flip state of a specific video stream on the input side.
Additionally, switch the singular test that was utilizing the rotation
metadata to instead override the input display rotation, thus leading
to the same result.
There is no MMX code for (add|put|put_signed)_pixels_clamped
since commit bfb28b5ce8, so use
declare_func instead of declare_func_emms() to also test that
we are not in MMX mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for diff_bytes since commit
230ea38de1, so use declare_func
instead of declare_func_emms() to also test that we are not
in MMX mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for add_int16 since commit
4b6ffc2880, so use declare_func
instead of declare_func_emms() to also test that we are not
in MMX mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for llviddsp after commit
fed07efcde, so use declare_func
instead of declare_func_emms() to also test that we are not
in MMX mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for pixblockdsp after commit
92b5800277, so use declare_func
instead of declare_func_emms() to also test that we are not
in MMX mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for audiodsp after commit
3d716d38ab, so use declare_func
instead of declare_func_emms() to also test that we are not
in MMX mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for blockdsp after commit
ee551a21dd, so use declare_func
instead of declare_func_emms() to also test that we are not
in MMX mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for vc1_inv_trans_8x8 or
vc1_unescape_buffer, so use declare_func instead of
declare_func_emms() to also test that we are not in MMX
mode after return.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no MMX code for loop filters since commit
6a551f1405, so use declare_func
instead of declare_func_emms() to also test that we are not
in MMX mode after return.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The data in SGI images is stored planar, so exporting
it via planar pixel formats is natural.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Unfortunately, it is common, and will remain so, that the Bit
manipulations are not enabled at compilation time. This is an official
policy for Debian ports in general (though they do not support RISC-V
officially as of yet) to stick to the minimal target baseline, which
does not include the B extension or even its Zbb subset.
For inline helpers (CPOP, REV8), compiler builtins (CTZ, CLZ) or
even plain C code (MIN, MAX, MINU, MAXU), run-time detection seems
impractical. But at least it can work for the byte-swap DSP functions.
This check is intended to be avoid buffer overflows,
yet there are four problems with it:
1. It has an in-built off-by-one error: len == out_end - out
is perfectly fine and nothing to worry about.
This off-by-one error led to the pixel in the lower-right corner
not being set properly for the back frame of the sample from
the rl2 FATE-test. This pixel is copied to every frame which
is the reason for the update to the reference file of said test.
With this patch, the output of the decoder matches the output
as captured from the reference decoder* (apart from the fact
that said reference somehow lacks the top part of the frame
(copied over from the background frame)).
2. Given that the stride of the buffer may be different
from the width of the video (despite one pixel taking one byte),
there is a second check lateron making the first check redundant
(if one returns immediately; a simple break at the second check
is not sufficient, because it only exits the inner loop).
3. The check is based around the assumption of the stride being
positive (it has this in common with the other check which
will be fixed in a future commit).
4. Even after fixing the off-by-one error, the check in
question is still triggered by all the non-background frames
in the FATE sample as well as by A1100100.RL2. In all these
cases, they use len == 255 and val == 128. For videos with
background frame this just means "copy from the background
frame", which would be done anyway lateron.* Yet for videos
without it copying it is necessary to avoid leaving
uninitialized parts in the video.
*: Available in https://samples.mplayerhq.hu/game-formats/voyeur-rl2/
**: Due to this, the code that copies the rest from the
back frame is no longer executed for any of the samples
available on the sample server. Given that these are only
the files from the demo version of this game, I don't know
whether this code is executed for any file in existence or not.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
RVV defines a total of 12 different extensions, including:
- 5 different instruction subsets:
- Zve32x: 8-, 16- and 32-bit integers,
- Zve32f: Zve32x plus single precision floats,
- Zve64x: Zve32x plus 64-bit integers,
- Zve64f: Zve32f plus Zve64x,
- Zve64d: Zve64f plus double precision floats.
- 6 different vector lengths:
- Zvl32b (embedded only),
- Zvl64b (embedded only),
- Zvl128b,
- Zvl256b,
- Zvl512b,
- Zvl1024b,
- and the V extension proper: equivalent to Zve64f and Zvl128b.
In total, there are 6 different possible sets of supported instructions
(including the empty set), but for convenience we allocate one bit for
each type sets: up-to-32-bit ints (RVV_I32), floats (RVV_F32),
64-bit ints (RVV_I64) and doubles (RVV_F64).
Whence the vector size is needed, it can be retrieved by reading the
unprivileged read-only vlenb CSR. This should probably be a separate
helper macro if needed at a later point.
This introduces compile-time and run-time CPU detection on RISC-V. In
practice, I doubt that FFmpeg will ever see a RISC-V CPU without all of
I, F and D extensions, and if it does, it probably won't have run-time
detection. So the flags are essentially always set.
But as things stand, checkasm wants them that way. Compare the ARMV8
flag on AArch64. We are nowhere near running short on CPU flag bits.
This also tests writing slice data in the unaligned mode
(some of these files use CAVLC) as well as updating
side data as well as parsing ISOBMFF avcc extradata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
~4x faster than the C version.
The shuffles in the 15pt dim1 are seriously expensive. Not happy with it,
but I'm contempt.
Can be easily converted to pure AVX by removing all vpermpd/vpermps
instructions.
Old one was written with the assumption only even inputs would be given.
This very messy replacement supports even and odd inputs, and supports
AVX2 for extra speed. The buffers given are usually quite big (4k samples),
so the speedup is worth it.
The new SSE version is still faster than the old inline asm version by 33%.
Also checkasm is provided to make sure this monstrosity works.
This fixes some FATE tests.
The aim of this test is to show the interleavement
of the file generated in the first pass; so make the
interleavement queue in the framecrc muxer in the second
pass as small as possible so that the framecrc muxer does not
fix wrong interleavement of the input file behind our backs.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
enc_dec is designed for raw input and output and computes
the PSNR between these two. The input of the shortest-sub
test is the idx file of a vobsub sub+idx combination
and the output is the output of framecrc of said vobsub
subtitle muxed into Matroska together with a synthesized
video. Calculating the PSNR between these two files makes
no sense, therefore switch to a transcode test, where
the ref file file contains the output of framecrc directly,
making the interleavement better visible in the ref file
at the cost of a larger ref file (>400 lines).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also covers writing mastering display metadata.
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Do this by setting AVCodecInternal.pad_samples.
This prevents reading into the frame's padding and writing
into the packet's padding.
This actually happened in our FATE tests (where the number of samples
is 2 mod 4), which therefore needed to be updated.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
APTX decodes four bytes of input to four stereo samples; APTX HD
does the same with six bytes of input. So it can be easily supported
in av_get_audio_frame_duration().
This fixes invalid durations and (derived) timestamps of demuxed
APTX HD packets and therefore fixed the timestamp in the aptx-hd
FATE test.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
We have de- and encoders for APTX and APTX HD, yet not FATE tests.
This commit therefore adds a transcoding test to utilize them.
Furthermore, during creating these tests it turned out that
the duration is set incorrectly for APTX HD. This will be fixed
in a future commit.
(Thanks to Andriy Gelman for finding an issue in an earlier version
that used a 192kHz input sample which does not work reliably accross
platforms.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Since introducing the various packed formats used by VAAPI (and p012),
we've noticed that there's actually a gap in how
av_find_best_pix_fmt_of_2 works. It doesn't actually assign any value
to having the same bit depth as the source format, when comparing
against formats with a higher bit depth. This usually doesn't matter,
because av_get_padded_bits_per_pixel() will account for it.
However, as many of these formats use padding internally, we find that
av_get_padded_bits_per_pixel() actually returns the same value for the
10 bit, 12 bit, 16 bit flavours, etc. In these tied situations, we end
up just picking the first of the two provided formats, even if the
second one should be preferred because it matches the actual bit depth.
This bug already existed if you tried to compare yuv420p10 against p016
and p010, for example, but it simply hadn't come up before so we never
noticed.
But now, we actually got a situation in the VAAPI VP9 decoder where it
offers both p010 and p012 because Profile 3 could be either depth and
ends up picking p012 for 10 bit content due to the ordering of the
testing.
In addition, in the process of testing the fix, I realised we have the
same gap when it comes to chroma subsampling - we do not favour a
format that has exactly the same subsampling vs one with less
subsampling when all else is equal.
To fix this, I'm introducing a small score penalty if the bit depth or
subsampling doesn't exactly match the source format. This will break
the tie in favour of the format with the exact match, but not offset
any of the other scoring penalties we already have.
I have added a set of tests around these formats which will fail
without this fix.
These tests test both the demuxer as well as the muxer
wherever possible. It is not always possible due to the fact
that the muxer supports more codecs than the demuxer.
The spdif demuxer does currently not set the need_parsing flag.
If one were to set this to AVSTREAM_PARSE_FULL, the test results
would change as follows:
- For spdif-aac-remux, the packets are currently padded to 16bits,
i.e. if the actual packet size is odd, there is a padding byte.
The parser splits this byte away into a one byte packet of its own.
Insanely, these one byte packets get the same duration as normal
packets, i.e. timing is ruined.
- The DCA-remux tests get proper duration/timestamps.
- In the spdif-mp2-remux test the demuxer marks the stream as
being MP2; the parser sets it to MP3 and this triggers
the "Codec change in IEC 61937" codepath; this test therefore
returns only two packets with the parser.
- For spdif-mp3-remux some bytes end up in different packets:
Some input packets of this file have an odd length (417B instead
of 418B like all the other packets) and are padded to 418B.
Without a parser, all returned packets from the spdif-demuxer
are 418B. With a parser, the packets that were originally 417B
are 417B again, but the padding byte has not been discarded,
but added to the next packet which is now 419B.
This fixes "Multiple frames in a packet" warning and avoids
an "Invalid data found when processing input" error when decoding.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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>
The function contains only two assignments, setting DVVideoContext.avctx
and AVCodecContext.chroma_sample_location. However, the decoder does not
use the former, and the encoder should not be setting the latter.
Therefore move the first assignment to dvenc and the second to dvdec.
Make the encoder warn if the user-signalled chroma sample location does
not match the supported one, and return an error on higher compliance
levels.
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.
This avoids triggering overflows in the filters, and avoids stray
test failures in the approximate functions on x86; due to rounding
differences, one implementation might overflow while another one
doesn't.
Signed-off-by: Martin Storsjö <martin@martin.st>
It is done later in ff_mpv_frame_start() (and nobody uses
current_picture_ptr between setting it in ff_mpv_frame_start()).
(The reason the vsynth*-h263-obmc ref files change is because
the call to ff_find_unused_picture() now happens after the older
pictures have been unreferenced in ff_mpv_frame_start(),
so that their slots in the picture array can be immediately
reused; the obmc code is somehow buggy and changes its output
depending on the earlier contents of the motion_val buffer.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Previously, the checkasm test always passed h=8, so no other cases
were tested.
Out of the me_cmp functions, in practice, some functions are hardcoded
to always assume a 8x8 block (ignoring the h parameter), while others
do use the parameter. For those with hardcoded height, both the
reference C function and the assembly implementations ignore the
parameter similarly.
The documentation for the functions indicate that heights between
w/2 and 2*w, within the range of 4 to 16, should be supported. This
patch just tests random heights in that range, without knowing what
width the current function actually uses.
Signed-off-by: Martin Storsjö <martin@martin.st>
Change the reference to exactly match the C reference in swscale,
instead of exactly matching the x86 SIMD implementations (which
differs slightly). Test with and without SWS_ACCURATE_RND - if this
flag isn't set, the output must match the C reference exactly,
otherwise it is allowed to be off by 2.
Mark a couple x86 functions as unavailable when SWS_ACCURATE_RND
is set - apparently this discrepancy hasn't been noticed in other
exact tests before.
Add a test for yuv2plane1.
Signed-off-by: Jonathan Swinney <jswinney@amazon.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
These tables are only used by encoders and only for the current picture;
ergo they need not be put into the picture at all, but rather into
the encoder's context. They also don't need to be refcounted,
because there is only one owner.
In contrast to this, the earlier code refcounts them which
incurs unnecessary overhead. These references are not unreferenced
in ff_mpeg_unref_picture() (they are kept in order to have something
like a buffer pool), so that several buffers are kept at the same
time, although only one is needed, thereby wasting memory.
The code also propagates references to other pictures not part of
the pictures array (namely the copy of the current/next/last picture
in the MpegEncContext which get references of their own). These
references are not unreferenced in ff_mpeg_unref_picture() (the
buffers are probably kept in order to have something like a pool),
yet if the current picture is a B-frame, it gets unreferenced
at the end of ff_mpv_encode_picture() and its slot in the picture
array will therefore be reused the next time; but the copy of the
current picture also still has its references and therefore
these buffers will be made duplicated in order to make them writable
in the next call to ff_mpv_encode_picture(). This is of course
unnecessary.
Finally, ff_find_unused_picture() is supposed to just return
any unused picture and the code is supposed to work with it;
yet for the vsynth*-mpeg4-adap tests the result depends upon
the content of these buffers; given that this patchset
changes the content of these buffers (the initial content is now
the state of these buffers after encoding the last frame;
before this patch the buffers used came from the last picture
that occupied the same slot in the picture array) their ref-files
needed to be changed. This points to a bug somewhere (if one removes
the initialization, one gets uninitialized reads in
adaptive_quantization in ratecontrol.c).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This codepath is enabled by default on arm, if the linux perf API
is available, unless disabled with --disable-linux-perf.
Signed-off-by: Martin Storsjö <martin@martin.st>
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".
The generated files are endian-dependent, so no checksums
may be part of the ref files.
Fixes ticket #9854.
Tested-by: Sebastian Ramacher <sramacher@debian.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The output file (even the filesize) of the recently added
EXR tests depends on the endianness; therefore checksums
of these files must not be part of the ref file. Therefore
this commit adds an option (unused for now) to disable these
checksums on a per-test basis.
In order to avoid having to check twice, the checksum and
the filesize info are moved to immediately follow one another;
this results into updates to the ref files of all lavf-image tests.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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>
It conflicts with the name of the test using the testtool
in libavformat.mak.
Fixes ticket #9841.
Reviewed-by: Pierre-Anthony Lemieux <pal@sandflow.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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.
For 422 frames we should not use hard coded 8 to calculate mb size for
uv plane. Chroma shift should be taken into consideration to be
compatiple with different sampling format.
The error is reported by fate test when av_cpu_max_align() return 64
on the platform supporting AVX512. This is a hidden error and it is
exposed after commit 17a59a634c.
mpeg2enc has a mechanism to reuse frames. When it computes SSE (sum of
squared error) on current mb, reconstructed mb will be wrote to the
previous mb space, so that the memory can be saved. However if the align
is 64, the frame is shared in somewhere else, so the frame cannot be
reused and a new frame to store reconstrued data is created. Because the
height of mb is wrong when compute sse on 422 frame, starting from the
second line of macro block, changed data is read when frame is reused
(we need to read row 16 rather than row 8 if frame is 422), and unchanged
data is read when frame is not reused (a new frame is created so the
original frame will not be changed).
That is why commit 17a59a634c exposes this
issue, because it add av_cpu_max_align() and this function return 64 on
platform supporting AVX512 which lead to creating a frame in mpeg2enc,
and this lead to the different outputs.
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
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>
- ff_pix_abs16_neon
- ff_pix_abs16_xy2_neon
In direct micro benchmarks of these ff functions verses their C implementations,
these functions performed as follows on AWS Graviton 3.
ff_pix_abs16_neon:
pix_abs_0_0_c: 141.1
pix_abs_0_0_neon: 19.6
ff_pix_abs16_xy2_neon:
pix_abs_0_3_c: 269.1
pix_abs_0_3_neon: 39.3
Tested with:
./tests/checkasm/checkasm --test=motion --bench --disable-linux-perf
Signed-off-by: Jonathan Swinney <jswinney@amazon.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
SSE3 instruction movdqa in ff_yuv2yuvX_sse3() expects a 16-byte aligned address for a memory address, or else a segfault is generated.
The src_pixels buffer below was not aligned to 16 bytes on the stack necessarily, so we got segfaults during fate-checkasm-sw_scale.
Therefore 16-byte align all of these local variables, aligning them too much shouldn't hurt.
- 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>
Use the md5 protocol instead of creating a file just to calculate
its MD5 checksum. This is possible because there are no output seeks
involved in any of these tests.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
log2() remains, this can either be replaced by a integer implementation or the table
hardcoded if needed
Tested-by: Anton Khirnov <anton@khirnov.net>
Tested-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The HEVC decoder can call these functions with smaller widths than the
functions themselves are designed to operate on so we should only check
the relevant output
Signed-off-by: J. Dekker <jdek@itanimul.li>
enc_dec() performs two ffmpeg runs - the first one encoding a source
file into a specified output format, the second one decoding previously
encoded file.
The arguments to this function currently have confusing names - e.g.
dec_opt contains _output_ (i.e. encoding) options for the second
(decoding) ffmpeg invocation. It is also possible to supply _input_
(i.e. decoding) options for the second ffmpeg run, but the argument
is currently unnamed and referred to by number.
Add an _in/_out suffix to argument names to make it clear what they are
used for. Give a name to input options for the decoding ffmpeg run.
