floating point precision will cause rgb*max generate different value on
x86_32 and x86_64. have pass fate test on x86_32 and x86_64 by using
lrintf to get the nearest integral value for rgb * max before av_clip.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Now we just use one ADTS raw frame to calculate the bit rate, it's
lead to a larger error when get the duration from bit rate, the
improvement cumulate Nth ADTS frames to get the average bit rate.
e,g used the command get the duration like:
ffprobe -show_entries format=duration -i fate-suite/aac/foo.aac
before this improvement dump the duration=2.173935
after this improvement dump the duration=1.979267
in fact, the real duration can be get by command like:
ffmpeg -i fate-suite/aac/foo.aac -f null /dev/null with time=00:00:01.97
Also update the fate-adtstoasc_ticket3715.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This is a requirement of the AV1-ISOBMFF spec. Section 2.1.
General Requirements & Brands states:
* It SHALL have the av01 brand among the compatible brands array of the FileTypeBox
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This causes regressions in end to end timestamps with mp3s and ffmpeg.
The revert is to avoid this regression in the 4.3 release
See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted.
This reverts commit 460132c998.
7546ac2fee made it so that the start_time for mp3 files is
adjusted for skip_samples. However, this appears incorrect because
subsequent packet timestamps are not adjusted and skip_samples are
applied by deleting data from a packet without changing the timestamp.
E.g., we are told the start_time is ~25ms and we get a packet with a
timestamp of 0 that has had the skip_samples discarded from it. As such
rendering engines may incorrectly discard everything prior to the
25ms thinking that is where playback should officially start. Since the
samples were deleted without adjusting timestamps though, the true
start_time is still 0.
Other formats like MP4 with edit lists will adjust both the start
time and the timestamps of subsequent packets to avoid this issue.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
A buffer whose size is not a multiple of four has been initialized using
consecutive writes of 32bits. This results in a stack-buffer-overflow
reported by ASAN in the checkasm-sw_scale FATE-test.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
changes since v1
- default behavior, no longer hidden behind decoder parameter
- updated tests to reflect change
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The Matroska muxer writes the Chapters early when chapters were already
available when writing the header; in this case any tags pertaining to
these chapters get written, too.
Yet if no chapters had been supplied before writing the header, Chapters
can also be written when writing the trailer if any are supplied. Tags
belonging to these chapters were up until now completely ignored.
This commit changes this: Writing the tags belonging to chapters has
been moved to mkv_write_chapters(). If mkv_write_tags() has not been
called yet (i.e. when chapters are written when writing the header),
the AVIOContext for writing the ordinary Tags element is used, but not
output, as this is left to mkv_write_tags() in order to only write one
Tags element. Yet if mkv_write_tags() has already been called,
mkv_write_chapters() will output a Tags element of its own which only
contains the tags for chapters.
When chapters are available initially, the corresponding tags will now
be the first tags in the Tags element; but the ordering of tags in Tags
is irrelevant anyway.
This commit also makes chapter_id_offset local to mkv_write_chapters()
as it is used only there and not reused at all.
Potentially writing a second Tags element means that the maximum number
of SeekHead entries had to be incremented. All the changes to FATE
result from the ensuing increase in the amount of space reserved for the
SeekHead (21 bytes more).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
We won't be able to seek back to write the actual duration anyway.
FATE-tests using the md5pipe command had to be updated due to this change.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Also fill x8-x17 with garbage before calling the function.
Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
We should just use a normal bl here, and the linker will add the 'x'
bit if necessary.
This fixes calling the checkasm_fail_func on windows, where the
code is built in thumb mode (and the linker doesn't clear the 'x'
bit in the blx instruction).
Signed-off-by: Martin Storsjö <martin@martin.st>
have tested on linux x86_32/64, mingw32/64 arm & mips qemu
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Tested on x86-32/64, mingw32/64, arm & mips qemu
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
This fixes tests on 32 bit x86 mingw with clang, which uses x87
fpu by default.
In this setup, while the get_expected function is declared to
return float, the compiler is (especially given the optimization
flags set) free to keep the intermediate values (in this case,
the return value from the inlined function) in higher precision.
This results in the situation where 7.28 (which actually, as
a float, ends up as 7.2800002098), multiplied by 100, is
728.000000 when really forced into a 32 bit float, but 728.000021
when kept with higher intermediate precision.
For the multiplication case, a more suitable epsilon would e.g.
be 2*FLT_EPSILON*fabs(expected_output), but just increase the
current hardcoded threshold for now.
Signed-off-by: Martin Storsjö <martin@martin.st>
Up until now, the Matroska muxer would mark a track as default if it had
the disposition AV_DISPOSITION_DEFAULT or if there was no track with
AV_DISPOSITION_DEFAULT set; in the latter case even more than one track
of a kind (audio, video, subtitles) was marked as default which is not
sensible.
This commit changes the logic used to mark tracks as default. There are
now three modes for this:
a) In the "infer" mode the first track of every type (audio, video,
subtitles) with default disposition set will be marked as default; if
there is no such track (for a given type), then the first track of this
type (if existing) will be marked as default. This behaviour is inspired
by mkvmerge. It ensures that the default flags will be set in a sensible
way even if the input comes from containers that lack the concept of
default flags. This mode is the default mode.
b) The "infer_no_subs" mode is similar to the "infer" mode; the
difference is that if no subtitle track with default disposition exists,
no subtitle track will be marked as default at all.
c) The "passthrough" mode: Here the track will be marked as default if
and only the corresponding input stream had disposition default.
This fixes ticket #8173 (the passthrough mode is ideal for this) as
well as ticket #8416 (the "infer_no_subs" mode leads to the desired
output).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Several EBML Master elements for which a good upper bound of the final
length was available were nevertheless written without giving an
upper bound of the final length to start_ebml_master(), so that their
length fields were eight bytes long. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer currently only adds CuePoints in three cases:
a) For video keyframes. b) For the first audio frame in a new Cluster if
in DASH-mode. c) For subtitles. This means that ordinary Matroska audio
files won't have any Cues which impedes seeking.
This commit changes this. For every track in a file without video track
it is checked and tracked whether a Cue entry has already been added
for said track for the current Cluster. This is used to add a Cue entry
for each first packet of each track in each Cluster.
Implements #3149.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Moreover, putting the Cues in front of the Clusters by reserving space
in advance is also tested.
The new capability of using ffprobe during a remux/transcode test are
used here for information about the chapters.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This is primarily intended to test that muxers correctly write chapters
or metadata; but given that it does this by having our demuxers read the
generated files, it also tests demuxers. And of course it may prove
useful for encoders, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, they were appended to the FATE_EXTERN-$(CONFIG_FFMPEG)
variable and were therefore activated when ffmpeg was enabled regardless
of whether ffprobe was enabled.
Also the same happened with FATE_SAMPLES_FASTSTART, although the
corresponding test (mov-faststart-4gb-overflow) only requires external
samples.
Furthermore, remove the unused FATE_FULL variable (FATE_EXTERN_FFPROBE has
taken its place).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Using random values for TrackUID and FileUID (as happens when the
AVFMT_FLAG_BITEXACT flag is not set) has the obvious downside of making
the output indeterministic. This commit mitigates this by writing the
potentially random values with a fixed size of eight byte, even if their
actual values would fit into less than eight bytes. This ensures that
even in non-bitexact mode, the differences between two files generated
with the same settings are restricted to a few bytes in the header.
(Namely the SegmentUID, the TrackUIDs (in Tracks as well as when
referencing them via TagTrackUID), the FileUIDs (in Attachments as
well as in TagAttachmentUID) as well as the CRC-32 checksums of the
Info, Tracks, Attachments and Tags level-1-elements.) Without this
patch, there might be an offset/a size difference between two such
files.
The FATE-tests had to be updated because the fixed-sized UIDs are also
used in bitexact mode.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If there are Attachments to write, the Matroska muxer currently
allocates two objects: An array that contains an entry for each
AttachedFile containing just the stream index of the corresponding
stream and the FileUID used for this AttachedFile; and a structure with
a pointer to said array and a counter for said array. These uids are
generated via code special to Attachments: It uses an AVLFG in the
normal and a sha of the attachment data in the bitexact case. (Said sha
requires an allocation, too.)
But now that an uid is generated for each stream in mkv_init(), there is
no need any more to use special code for generating the FileUIDs of
AttachedFiles: One can simply use the uid already generated for the
corresponding stream. And this makes the whole allocations of the
structures for AttachedFiles as well as the structures itself superfluous.
They have been removed.
In case AVFMT_FLAG_BITEXACT is set, the uids will be different from the
old ones which is the reason why the FATE-test lavf-mkv_attachment
needed to be updated. The old method had the drawback that two
AttachedFiles with the same data would have the same FileUID.
The new one doesn't.
Also notice that the dynamic buffer used to write the Attachments leaks
if an error happens when writing the buffer. By removing the
allocations potential sources of errors have been removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Tags in the Matroska file format can be summarized as follows: There is
a level 1-element called Tags containing one or many Tag elements each
of which in turn contain a Targets element and one or many SimpleTags.
Each SimpleTag roughly corresponds to a single key-value pair similar to
an AVDictionaryEntry. The Targets meanwhile contains information to what
the metadata contained in the SimpleTags contained in the containing Tag
applies (i.e. to the file as a whole or to an individual track).
The Matroska muxer writes such metadata. It puts the metadata of every
stream into a Tag whose Targets makes it point to the corresponding
track. And if the output is seekable, then it also adds another Tag for
each track whose Targets corresponds to the track and where it reserves
space in a SimpleTag to write the duration at the end of the muxing
process into.
Yet there is no reason to write two Tag elements for a track and a few
bytes (typically 24 bytes per track) can be saved by adding the duration
SimpleTag to the other Tag of the same track (if it exists).
FATE has been updated because the output files changed. (Tests that
write to unseekable output (pipes) needn't be updated (no duration tag
has ever been written for them) and the same applies to tests without
further metadata.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It represents the relationship between them more naturally and will be
useful in the following commits.
Allows significantly more frames in fate-h264-attachment-631 to be
decoded.
containing updated extradata, in this case a new FLAC streaminfo.
Furthermore, it also tests that the Matroska muxer is able to preserve
uncommon channel layouts by adding Vorbis comments to the CodecPrivate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It might be used by the Matroska muxer. This is also the reason why the
FATE-tests for muxing WavPack into Matroska needed to be updated: They
now write the correct version 4.07 and not 4.03 as before.
Reviewed-by: David Bryant <david@wavpack.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
mkvmerge versions 6.2 to 40.0 had a bug that made it not propagate the
WavPack extradata (containing the WavPack version) during remuxing from
a Matroska file; currently our demuxer would treat every WavPack block
encountered as invalid data (unless the WavPack stream is to be
discarded (i.e. the streams discard is >= AVDISCARD_ALL)) and try to
resync to the next level 1 element.
Luckily, the WavPack version is currently not really important; so we
fix this problem by assuming a version. David Bryant, the creator of
WavPack, recommended using version 0x410 (the most recent version) for
this. And this is what this commit does.
A FATE-test for this has been added.
Reviewed-by: David Bryant <david@wavpack.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Add overflow test for hevc_add_res when int16_t coeff = -32768.
The result of C is good, while ASM is not.
To verify:
make fate-checkasm-hevc_add_res
ffmpeg/tests/checkasm/checkasm --test=hevc_add_res
./checkasm --test=hevc_add_res
checkasm: using random seed 679391863
MMXEXT:
hevc_add_res_4x4_8_mmxext (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
SSE2:
hevc_add_res_8x8_8_sse2 (hevc_add_res.c:69)
hevc_add_res_16x16_8_sse2 (hevc_add_res.c:69)
hevc_add_res_32x32_8_sse2 (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
AVX:
hevc_add_res_8x8_8_avx (hevc_add_res.c:69)
hevc_add_res_16x16_8_avx (hevc_add_res.c:69)
hevc_add_res_32x32_8_avx (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
AVX2:
hevc_add_res_32x32_8_avx2 (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
checkasm: 8 of 14 tests have failed
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
check_func will return NULL for functions that have already been tested. If
the func is tested and skipped (which happens several times), there is no
need to prepare data(randomize_buffers and memcpy).
Move relative code in compare_add_res(), prepare data and do check only if
the function is not tested.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Up until e7ddafd5, the Matroska muxer wrote two SeekHeads: One at the
beginning referencing the main level 1 elements (i.e. not the Clusters)
and one at the end, referencing the Clusters. This second SeekHead was
useless and has therefore been removed. Yet the SeekHead-related
functions and structures are still geared towards this usecase: They
are built around an allocated array of variable size that gets
reallocated every time an element is added to it although the maximum
number of Seek entries is a small compile-time constant, so that one should
rather include the array in the SeekHead structure itself; and said
structure should be contained in the MatroskaMuxContext instead of being
allocated separately.
The earlier code reserved space for a SeekHead with 10 entries, although
we currently write at most 6. Reducing said number implied that every
Matroska/Webm file will be 84 bytes smaller and required to adapt
several FATE tests; furthermore, the reserved amount overestimated the
amount needed for for the SeekHead's length field and how many bytes
need to be reserved to write a EBML Void element, bringing the total
reduction to 89 bytes.
This also fixes a potential segfault: If !mkv->is_live and if the
AVIOContext is initially unseekable when writing the header, the
SeekHead is already written when writing the header and this used to
free the SeekHead-related structures that have been allocated. But if
the AVIOContext happens to be seekable when writing the trailer, it will
be attempted to write the SeekHead again which will lead to segfaults
because the corresponding structures have already been freed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The WebM DASH Manifest muxer can write manifests for live streams and
these contain an entry that depends on the time the manifest is written;
an AVOption to make the output reproducible has been added for tests.
But this is unnecessary, as there already is a method for reproducible
output: The AVFMT_FLAG_BITEXACT-flag of the AVFormatContext. Therefore
this commit removes the custom option.
Given that the description of said option contained "private option -
users should never set this" and that it was not documented in
muxers.texi, no deprecation period for this option seemed necessary.
The commands of the FATE-tests for this muxer have been changed to no
longer use this option.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This fixes mpeg2video stream copies to mpeg muxer like this:
ffmpeg -i xdcamhd.mxf -c:v copy output.mpg
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Utilizes a subpicture sample with one decodable subpicture for the
test.
Based on a failing test case in reported by Michael in
https://ffmpeg.org/pipermail/ffmpeg-devel/2019-February/240398.html
which at the time had no test case for it.
Additionally, this is the first test case for the presentation
graphics format.
According to the H.264 specifications, the only NAL units that need to
have four byte startcodes in H.264 Annex B format are SPS/PPS units and
units that start a new access unit. Before af7e953a, the first of these
conditions wasn't upheld as already existing in-band parameter sets
would not automatically be written with a four byte startcode, but only
when they already were at the beginning of their input packets. But it
made four byte startcodes be used too often as every unit that is written
together with a parameter set that is inserted from extradata received a
four byte startcode although a three byte start code would suffice
unless the unit itself were a parameter set.
FATE has been updated to reflect the changes. Although the patch leaves
the extradata unchanged, the size of the extradata according to the FATE
reports changes. This is due to a quirk in ff_h2645_packet_split which
is used by extract_extradata: If the input is Annex B, the first zero of
a four byte startcode is considered a part of the last unit (if any).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The standard does not seem to require the counter to be zero based, but some
checker tools (MyriadBits MXFInspect, Interra Baton) have validations against 0
start...
Fixes ticket #6781.
Signed-off-by: Marton Balint <cus@passwd.hu>
RFC 3986 states that the generic syntax uses the slash ("/"), question mark
("?"), and number sign ("#") characters to delimit components that are
significant to the generic parser's hierarchical interpretation of an
identifier.
Signed-off-by: Marton Balint <cus@passwd.hu>
The tests for concat use this option which is scheduled for removal and
does nothing any more. So remove it; otherwise, these tests would fail
at the next major version bump.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
When a Matroska Block is only stored in compressed form, the size of
the uncompressed block is not explicitly coded and therefore not known
before decompressing it. Therefore the demuxer uses a guess for the
uncompressed size: The first guess is three times the compressed size
and if this is not enough, it is repeatedly incremented by a factor of
three. But when this happens with lzo, the decompression is neither
resumed nor started again. Instead when av_lzo1x_decode indicates that x
bytes of input data could not be decoded, because the output buffer is
already full, the first (not the last) x bytes of the input buffer are
resent for decoding in the next try; they overwrite already decoded
data.
This commit fixes this by instead restarting the decompression anew,
just with a bigger buffer.
This seems to be a regression since 935ec5a1.
A FATE-test for this has been added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This test tests that demuxing ProRes that is muxed like it should be in
Matroska (i.e. with the first header ("icpf") atom stripped away) works;
it also tests bz2 decompression as well as the handling of
unknown-length clusters.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Up until now, the microdvd demuxer uses av_strdup() to allocate the
extradata from a string; its length is set to strlen() + 1, i.e.
including the \0 at the end. Upon remuxing, the muxer would simply copy
the extradata at the beginning, including the \0.
This commit changes this by not adding the \0 to the size of the
extradata; the muxer now delimits extradata by inserting a \n. This
required to change the subtitles-microdvd-remux FATE-test.
Furthermore, the extradata is now allocated with zeroed padding.
The microdvd decoder is not affected by this, as it didn't use the size
of the extradata at all, but treated it as a C-string.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The stereo_interpolate functions add h_step to the values h
BUF_SIZE times. Within the stereo_interpolate C functions, the
values h (h0-h3, h00-h13) are declared as local float variables,
but the compiler is free to keep them in a register with extra
precision.
If the accumulation is rounded to 32 bit float precision after
each step, the less significant bits of h_step end up ignored
and the sum can deviate, affecting the end result more than
the currently set EPS.
By clearing the log2(BUF_SIZE) lower bits of h_step, we make sure
that the accumulation shouldn't differ significantly, regardless
of any extra precision in the accmulating register/variable.
This fixes the aacpsdsp checkasm test when built with clang for
mingw/x86_32.
Signed-off-by: Martin Storsjö <martin@martin.st>
In these cases, we must pass the full path of the file to ffprobe
(as the current working dir on the remote system, e.g. when invoked
with "ssh remote ffprobe ..." isn't the wanted one).
The input filename passed to ffprobe is also included in the output,
which is part of the reference test data. Add a new option to
ffprobe to allow overriding what path is printed, to keep the
original relative path in the tests.
An alternative approach could be an option to allow requesting omitting
the file name from the dumped data, and updating the test references
accordingly.
Signed-off-by: Martin Storsjö <martin@martin.st>
5 cabac states for cbf_cb and cbf_cr are supported according to
Table 9-4.
Add a test for 64x64 4:4:4 8bit HEVC clips with TUDepth = 4, cbf_cr > 0.
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
When testing on a memory limited system, these tests consume a
significant amount of memory and can often fail if testing by running
multiple processes in parallel.
Signed-off-by: Martin Storsjö <martin@martin.st>
The IVF muxer autoinserts the av1_metadata filter unconditionally, which is
not desirable for these tests.
Signed-off-by: James Almer <jamrial@gmail.com>
As the values generated by av_bmg_get can be arbitrarily large
(only the stddev is specified), we can't use a fixed tolerance.
Calculate a dynamic tolerance (like in float_dsp from 38f966b222),
based on the individual steps of the calculation.
This fixes running this test with certain seeds, when built with
clang for mingw/x86_32.
Signed-off-by: Martin Storsjö <martin@martin.st>
These dependencies are evaluted by make and must be expressed with
the paths as in the local filesystem.
Signed-off-by: Martin Storsjö <martin@martin.st>
The tremolo filter uses floating point internally, and uses
multiplication factors derived from sin(fmod()), neither of
which is bitexact for use with framecrc.
This fixes running this test when built with for mingw/x86_32
with clang.
In this case, a 1 ulp difference in the output from fmod() would
end up in an output from the filter that differs by 1 ulp, but
which makes the lrint() in swresample/audioconvert.c round in a
different direction.
Signed-off-by: Martin Storsjö <martin@martin.st>
As the values generated by av_bmg_get can be arbitrarily large
(only the stddev is specified), we can't use a fixed tolerance.
This matches what was done for test_vector_dmul_scalar in
38f966b222.
This fixes the float_dsp checkasm test for some seeds, when built
with clang for mingw/x86_32.
Signed-off-by: Martin Storsjö <martin@martin.st>
contained in Vorbis comments in the CodecPrivate of flac tracks.
Moreover, it also tests header removal compression.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This test contains a track with zlib compressed CodecPrivate in addition
to compressed frames; the former was unchecked before.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: fate-fitsdec-bitpix-64
Possibly Fixes: -nan is outside the range of representable values of type 'unsigned short'
Possibly Fixes: 17769/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FITS_fuzzer-5678314672357376
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Unlike other tf.*.conv2d layers, tf.nn.conv2d does not create many
nodes (within a scope) in the graph, it just acts like other layers.
tf.nn.conv2d only creates one node in the graph, and no internal
nodes such as 'kernel' are created.
The format of native model file is also changed, a flag named
has_bias is added, so change the version number.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
Allows the creation of the sdtp atom while remuxing MP4 to MP4. This
atom is required by Apple devices (iPhone, Apple TV) in order to accept
2160p medias.
The flac parser uses a fifo to buffer its data. Consequently, when
searching for sync codes of flac packets, one needs to take care of
the possibility of wraparound. This is done by using an optimized start
code search that works on each of the continuous buffers separately and
by explicitly checking whether the last pre-wrap byte and the first
post-wrap byte constitute a valid sync code.
Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched
for (the start of) a sync code because a header that might be found in this
region might not be completely available. These bytes ought to be searched
lateron when more data is available or when flushing.
Unfortunately there was an off-by-one error in the calculation of the
length to search of the post-wrap buffer: It was too large, because the
calculation was based on the amount of bytes available in the fifo from
the last pre-wrap byte onwards. This meant that a header might be
parsed twice (once prematurely and once regularly when more data is
available); it could also mean that an invalid header will be treated as
valid (namely if the length of said invalid header is
MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the
last byte of this potential header happens to be the right CRC-8).
Should a header be parsed twice, the second instance will be the best child
of the first instance; the first instance's score will be
FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than
the second instance's score. So the frame belonging to the first
instance will be output and it will be done as a zero length frame (the
difference of the header's offset and the child's offset). This has
serious consequences when flushing, as returning a zero length buffer
signals to the caller that no more data will be output; consequently the
last frames not yet output will be dropped.
Furthermore, a "sample/frame number mismatch in adjacent frames" warning
got output when returning the zero-length frame belonging to the first
header, because the child's sample/frame number of course didn't match
the expected sample frame/number given its parent.
filter/hdcd-mix.flac from the FATE-suite was affected by this (the last
frame was omitted) which is the reason why several FATE-tests needed to
be updated.
Fixes ticket #5937.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
A threshold of 1 is sufficient for simple_dump_cut.webm, 10 is used
just to be sure the next truncated file doesnt cause the same issue
Obvious alternative fixes are to simply accept that the file is broken or to
write some advanced error concealment or to
simply accept that the decoder wont stop at the end of input.
Fixes: Ticket 8069 (artifacts not the differing md5 which was there before 1afd246960)
Fixes: simple_dump_cut.webm
Fixes: regression of 1afd246960
fate-vp5 changes because the last frame is truncated and now handled
differently.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Because the lavf_container is sometimes called with only 2 arguments,
fate tests produce bash errors like this:
tests/fate-run.sh: 299: test: =: unexpected operator
This commit fixes this.
Reviewed-by: Limin Wang <lance.lmwang@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Right now, the concat filter does not set the frame_rate value on any of
the out links. As a result, the default ffmpeg behaviour kicks in - to
copy the framerate from the first input to the outputs.
If a later input is higher framerate, this results in dropped frames; if
a later input is lower framerate it might cause judder.
This patch checks if all of the video inputs have the same framerate, and
if not it sets the out link to use '1/0' as the frame rate, the value
meaning "unknown/vfr".
A test is added to verify the VFR behaviour. The existing test for CFR
behaviour passes unchanged.
the info can be saved in dnn operand object without regenerating again and again,
and it is also needed for layer split/merge, and for memory reuse.
to make things step by step, this patch just focuses on c code,
the change within python script will be added later.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
This makes the code bitexact between platforms.
Intermediate timestamps between frames are preserved.
The timebase is simplified.
Rounding differs from doubles in cases where timestamps/durations
are "funny"
Suggested-by: jb
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This reverts commit a9dacdeea6.
This patch effectively made the decoder output vfr content out of samples
where cfr is expected.
Addresses ticket #7880.
Signed-off-by: James Almer <jamrial@gmail.com>
The packet counting based approach caused excessive sdt/pat/pmt for VBR, so
let's use a timestamp based approach instead similar to how we emit PCRs.
SDT/PAT/PMT period should be consistent for both VBR and CBR from now on.
Also change the type of sdt_period and pat_period to AV_OPT_TYPE_DURATION so no
floating point math is necessary.
Fixes ticket #3714.
Signed-off-by: Marton Balint <cus@passwd.hu>
background:
DNN (deep neural network) is a sub module of libavfilter, and FATE/dnn
is unit test for the DNN module, one unit test for one dnn layer.
The unit tests are not based on the APIs exported by libavfilter,
they just directly call into the functions within DNN submodule.
There is an issue when run the following command:
build$ ../ffmpeg/configure --disable-static --enable-shared
make
make fate-dnn-layer-pad
And part of error message:
tests/dnn/dnn-layer-pad-test.o: In function `test_with_mode_symmetric':
/work/media/ffmpeg/build/src/tests/dnn/dnn-layer-pad-test.c:73: undefined reference to `dnn_execute_layer_pad'
The root cause is that function dnn_execute_layer_pad is a LOCAL symbol
in libavfilter.so, and so the linker could not find it when build dnn-layer-pad-test.
To check it, just run: readelf -s libavfilter/libavfilter.so | grep dnn
So, add dependency in fate/dnn Makefile with ffmpeg static libraries.
This is the same method used in fate/checkasm
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
This fixes make fate issue for frame thread scale in my local testing
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
At the moment scene change detection score uses all planes to detect scene
changes. In this regard this is similar how the frozen frames detection works.
However, in classic encoding scene change detection typically only uses the Y
plane.
We might get more resonable scores for scene change if we also use only
the Y plane for calculating the score if the pixel format is YUV. Although
this will require additional work once packed YUV formats are added,
because for the moment the generic scene sad score calculation has no way
to ignore some components in a packed format.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
cuda_runtime.h as well as dynlink_loader.h used nonstandard inclusion
guards with an AV_ prefix, although these files are not in an libav*/
path. So change the inclusion guards and adapt the ref file of the
source fate test accordingly.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
why change .4 to .25, it's for:
one scenecut(pkt_pts=20040) isn't detected by 0.4 threshold
why not change to 0.3 instead of 0.25:
it will miss the scenecut(pkt_pts=20040) after applying the next
patch which enables yuvj420
for fate testing, it's better to catch all scenecut scenes.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
