This together with adjusting the inclusion define allows for the
build to not fail with latest Vulkan-Headers that contain the
stabilized Vulkan AV1 decoding definitions.
Compilation fails currently as the AV1 header is getting included
via hwcontext_vulkan.h -> <vulkan/vulkan.h> -> vulkan_core.h, which
finally includes vk_video/vulkan_video_codec_av1std.h and the decode
header, leading to the bundled header to never defining anything
due to the inclusion define being the same.
This fix is imperfect, as it leads to additional re-definition
warnings for things such as
VK_STD_VULKAN_VIDEO_CODEC_AV1_DECODE_SPEC_VERSION. , but it is
not clear how to otherwise have the bundled version trump the
actually standardized one for a short-term compilation fix.
Currently, this only affects untagged RGB/XYZ/Gray, which get forced to
their corresponding metadata before entering the filter graph. The main
justification for this change, however, is the planned ability to add
automatic promotion of unspecified yuv to mpeg range yuv.
Notably, this change will never allow accidentally cross-promoting
unspecified to jpeg or to a specific YUV matrix, since that is still
bound by the constraints of YUV range negotiation as set up by
query_formats.
Since 7bf1b9b357,
the test produces ordinary \n, yet this is not what the reference
file used for the most time, leading to test failures.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Previously, we produced output with either \r\n or mixed line endings.
This was undesirable unto itself, but also made working with patches affecting
FATE output particularly challenging, especially via the mailing list.
Everything that consumes the SSA/ASS format is line-ending-agnostic,
so \n is selected to simplify git/ML usage in FATE.
Extra \r characters at the end of a packet are dropped. These are always
ignored by the renderer anyway.
The previous assumption that DXV needs to be aligned to 16x16 was
erroneous. 4x4 works just as well, and FATE decoder tests pass for all
texture formats.
On the encoder side, we should reject input that isn't 4x4 aligned,
like the HAP encoder does, and stop aligning to 16x16. This both solves
the uninitialized reads causing current FATE tests to fail and produces
smaller encoded outputs.
With regard to correctness, I've checked the decoding path by encoding a
real-world sample with git master, and decoding it with
ffmpeg -i dxt1-master.mov -c:v rawvideo -f framecrc -
The results are exactly the same between master and this patch.
On the encoding side, I've encoded a real-world sample with both master
and this patch, and decoded both versions with
ffmpeg -i dxt1-{master,patch}.mov -c:v rawvideo -f framecrc -
Under this patch, results for both inputs are exactly the same.
In other words, the extra padding gained by 16x16 alignment over 4x4
alignment has no impact on decoded video.
Signed-off-by: Connor Worley <connorbworley@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes all ff_draw_* based filters aware of YUV colorspaces and
ranges. Needed for YUVJ removal. Also fixes a bug where e.g. vf_pad
would generate a limited range background even after conversion to
full-scale grayscale.
The FATE changes were a consequence of the aforementioned bugfix - the
gray scale files are output as full range (due to conversion by
libswscale, which hard-codes gray = full), and appropriately tagged as
such, but before this change the padded version incorrectly used
a limited range (16) black background for these formats.
Use 8 packets/frames by default rather than 1, which seems to provide
better throughput.
Allow -thread_queue_size to set the muxer queue size manually again.
Finishes fixing vp5/potter512-400-partial.avi
The fate-matroska-ms-mode test ref is updated to reflect that the Speex decoder
can now read the stream.
Signed-off-by: James Almer <jamrial@gmail.com>
Covers muxing from raw pcm audio input into FLAC, using several scalable layouts,
and demuxing the result.
Signed-off-by: James Almer <jamrial@gmail.com>
This is the 64bit version of Chris Doty-Humphreys SFC64
Compared to the LCGs these produce much better quality numbers.
Compared to LFGs this needs less state. (our LFG has 224 byte
state for its 32bit version) this has 32byte state
Also the initialization for our LFG is slower.
This is also much faster than KISS or PCG.
This commit replaces the broken LCG used before.
(broken as it had only a period ~200M due to being put in a double)
This changes the output from random() which is why libswresample.mak
is updated, update was done using the command in libswresample.mak
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
On some platforms (in particular, ARM/AArch64), the implementation
of AV_READ_TIME() may use a privileged instruction - in such
cases, benchmarking just fails with a SIGILL.
Instead of crashing, try executing AV_READ_TIME() once within
a region with the signal handler active, to allow gracefully
informing the user about the issue.
This matches the dav1d checkasm commit
95a192549a448b70d9542e840c4e34b60d09b093.
Signed-off-by: Martin Storsjö <martin@martin.st>
Parse iprp and iinf boxes and its child boxes to get the actual codec used
(AV1 for avif, HEVC for heic), and properly export extradata and other
properties in a generic way.
The avif tests reference files are updated as the extradata is now exported.
Based on a patch by Swaraj Hota
Co-authored-by: Swaraj Hota <swarajhota353@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Marth64 <marth64@proxyid.net>
Raw Captions With Time (RCWT) is a format native to ccextractor, a commonly
used open source tool for processing 608/708 closed caption (CC) sources.
It can be used to archive the original, raw CC bitstream and to produce
a source file file for later CC processing or conversion. As a result,
it also allows for interopability with ccextractor for processing CC data
extracted via ffmpeg. The format is simple to parse and can be used
to retain all lines and variants of CC.
A free specification of RCWT can be found here:
https://github.com/CCExtractor/ccextractor/blob/master/docs/BINARY_FILE_FORMAT.TXT
This muxer implements the specification as of 01/05/2024, which has
been stable and unchanged for 10 years as of this writing.
This muxer will have some nuances from the way that ccextractor muxes RCWT.
No compatibility issues when processing the output with ccextractor
have been observed as a result of this so far, but mileage may vary
and outputs will not be a bit-exact match.
Specifically, the differences are:
(1) This muxer will identify as "FF" as the writing program identifier, so
as to be honest about the output's origin.
(2) ffmpeg's MPEG-1/2, H264, HEVC, etc. decoders extract closed captioning
data differently than ccextractor from embedded SEI/user data.
For example, DVD captioning bytes will be translated to ATSC A53 format.
This allows ffmpeg to handle 608/708 in a consistant way downstream.
This is a lossless conversion and the meaningful data is retained.
(3) This muxer will not alter the extracted data except to remove invalid
packets in between valid CC blocks. On the other hand, ccextractor
will by default remove mid-stream padding, and add padding at the end
of the stream (in order to convey the end time of the source video).
This replaces the riscv specific handling from
7212466e73 (which essentially is
reverted), with a different implementation of the same (plus a bit
more), based on the corresponding feature in dav1d's checkasm,
supporting both Unix and Windows.
See in particular the dav1d commits
0b6ee30eab2400e4f85b735ad29a68a842c34e21,
0421f787ea592fd2cc74c887f20b8dc31393788b,
8501a4b20135f93a4c3b426468e2240e872949c5 and
d23e87f7aee26ddcf5f7a2e185112031477599a7, authored by Henrik Gramner.
The overall approach compared to the existing implementation for
riscv is the same; set up a signal handler, store the state with
sigsetjmp, jump out of the crashing function with siglongjmp.
The main difference is in what happens when the signal handler
is invoked. In the previous implementation, it would resume from
right before calling the crashing function, and then skip that call
based on the setjmp return value.
In the imported implementation from dav1d, we return to right before
the check_func() call, which will skip testing the current function
(as the pointer is the same as it was before).
Other differences are:
- Support for other signal handling mechanisms (Windows
AddVectoredExceptionHandler)
- Using RtlCaptureContext/RtlRestoreContext instead of setjmp/longjmp
on Windows with SEH
- Only catching signals once per function - if more than one
signal is delivered before signal handling is reenabled, any
signal is handled as it would without our handler
- Not using an arch specific signal handler written in assembly
Signed-off-by: Martin Storsjö <martin@martin.st>
The layout for the frame flags is as follow:
chroma_format u(2)
reserved u(2)
interlace_mode u(2)
reserved u(2)
chroma_format has 2 allowed values:
0: reserved
1: reserved
2: 4:2:2
3: 4:4:4
interlace_mode has 3 allowed values:
0: progressive
1: tff
2: bff
3: reserved
0x80 is what we expect for "422 not interlaced", and the extra 0x2 from
0x82 is actually writing into the reserved bits.
This byte represents 4 reserved bits followed by 4 alpha_channel_type bits.
alpha_channel_type currently has 3 differents defined values: 0 (no
alpha), 1 (8b alpha), and 2 (16b alpha), all the other values are
reserved. The 4 initial reserved bits are expected to be 0.
This byte represents 4 reserved bits followed by 4 alpha_channel_type bits.
alpha_channel_type currently has 3 differents defined values: 0 (no
alpha), 1 (8b alpha), and 2 (16b alpha), all the other values are
reserved. This part is correctly written (alpha_bits>>3 does the correct
thing), but the 4 initial bits are reserved.
Quoting SMPTE RDD 36:2015:
A decoder shall abort if it encounters a bitstream with an unsupported
bitstream_version value. If 0, the value of the chroma_format syntax
element shall be 2 (4:2:2 sampling) and the value of the
alpha_channel_type element shall be 0 (no encoded alpha); if 1, any
permissible value may be used for those syntax elements.
So if we're not in 4:2:2 or if there is alpha, we are not allowed to use
version 0.
They are similar to AVIF images (both use the HEIF container).
The only additional work needed is to parse the hvcC box and put
it in the extradata.
With this patch applied, ffmpeg (when built with an HEVC decoder)
is able to decode the files in
https://github.com/nokiatech/heif/tree/gh-pages/content/images
Also add a couple of fate tests with samples from
https://github.com/nokiatech/heif_conformance/tree/master/conformance_files
Partially fixes trac ticket #6521.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The start code is matched against 0x000001, zero_byte was treated
as last byte of last frame rather than the beginning of next frame.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Motivated by YUVJ removal. This change will allow full negotiation
between color ranges and matrices as needed. By default, all ranges and
matrices are marked as supported.
Because grayscale formats are currently handled very inconsistently (and
in particular, assumed as forced full-range by swscale), we exclude them
from negotiation altogether for the time being, to get this API merged.
After filter negotiation is available, we can relax the
grayscale-is-forced-jpeg restriction again, when it will be more
feasible to do so without breaking a million test cases.
Note that this commit updates one FATE test as a consequence of the
sanity fallback for non-YUV formats. In particular, the test case now
writes rgb24(pc, gbr/unspecified/unspecified) to the matroska file,
instead of rgb24(unspecified/unspecified/unspecified) as before.
For memcpy and memcmp, we need to multiply by the element size,
otherwise we're copying and comparing only a fraction of the buffer.
For decorrelate_sr, the buffer p1 is the one that is mutated;
copy and check p1 instead of p2.
For decorrelate_sm, both buffers are mutated, so copy and check
both of them.
For decorrelate_sm, the memcpy initialization of p1 and p1_2 was
reversed - p1 is filled with randomize, but then memcpy copies from
p1_2 to p1. As p1_2 is uninitialized at this point, clang concluded
that the copy was bogus and omitted it entirely, triggering failures
in this test on x86 (where there was an existing assembly implementation
to test).
Signed-off-by: Martin Storsjö <martin@martin.st>
The ffmpeg coding style doesn't usually use const on scalar
parameters (or on the pointer values - as opposed to the type
that is pointed to, where it has a semantic meaning), contrary
to the dav1d coding style (where this was imported from).
This avoids warnings about differences in the type signatures
between declaration and definition of this function, with older
versions of MSVC.
The issue was observed with one version of MSVC 2017,
19.16.27024.1, with warnings like these:
src/tests/checkasm/checkasm.c(969): warning C4028: formal parameter 3 different from declaration
The warning itself is bogus as the const here is harmless, and
newer versions of MSVC no longer warn about this.
Signed-off-by: Martin Storsjö <martin@martin.st>
This can be used to run tests multple times, with e.g. differing
QEMU settings, by adding something like this to the FATE configuration
file:
target_exec="qemu-aarch64-static"
fate_targets="fate-checkasm fate-cpu"
fate_environments="sve128 sve256 sve512"
sve128_env="QEMU_CPU=max,sve128=on"
sve256_env="QEMU_CPU=max,sve256=on"
sve512_env="QEMU_CPU=max,sve512=on"
It's also possible to customize the target_exec command further
by injecting a sufficiently quoted variable into it, which then can
be updated for each run, e.g. target_exec="\$(CUR_EXEC_CMD)".
For each of the environment names in fate_environments, the tests
that are run get the name suffixed on the fate tests in the
test log and fate report, e.g. "fate-checkasm-h264dsp_sve128".
Signed-off-by: Martin Storsjö <martin@martin.st>
This can be useful if doing testing of uncommon CPU extensions by
running tests with QEMU (by configuring with e.g.
"target_exec=qemu-aarch64"), by only running the checkasm tests,
to get a reasonable test coverage without excessive test runtime.
For such a config, setting fate_targets="fate-checkasm fate-cpu"
can be a good tradeoff.
Signed-off-by: Martin Storsjö <martin@martin.st>
During a resampling operation where
1) user has specified first_pts
2) SWR_FLAG_RESAMPLE is not set initially (directly or otherwise)
3) first_pts has been fulfilled (always using hard compensation)
then upon first encountering a delay where a soft compensation is
required, swr_set_compensation will lead to another init of swr which
will reset outpts to the specified firstpts thus leading to an output
frame having its pts = firstpts. When the next input frame is received,
swr will see a large delay and inject silence from firstpts to the
current frame's pts. This can lead to severe desync and in worst case,
loss of audio playback.
Parameter firstpts initialized to AV_NOPTS_VALUE in swr_alloc and then
checked in swr_init to avoid resetting outpts, thus avoiding reapplication
of firstpts.
Fixes#4131.
This test verifies the parser's handling of multiframe JXL files that
have an entropy-encoded permuted table of contents for each frame. The
testcase is actually six JXL codestreams concatenated together, and the
parser needs to be able to find the boundaries.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
- Fixes YA formats, because previous code always assumed alpha as the 4th
component.
- Fixes PAL format (as long as 0 is black, as in a systematic palette), because
previous code assumed it as limited Y.
- Fixes XYZ format because it does not need nonzero chroma components
- Fixes xv30be as the bitstream mode got merged to the non-bitstream mode.
Signed-off-by: Marton Balint <cus@passwd.hu>
Change the main loop and every component (demuxers, decoders, filters,
encoders, muxers) to use the previously added transcode scheduler. Every
instance of every such component was already running in a separate
thread, but now they can actually run in parallel.
Changes the results of ffmpeg-fix_sub_duration_heartbeat - tested by
JEEB to be more correct and deterministic.
Avoid making decisions based on current graph input state, which makes
the output dependent on the order in which the frames from different
inputs are interleaved.
Makes the output of fate-filter-overlay-dvdsub-2397 more correct - the
subtitle appears two frames later, which is closer to its PTS as stored
in the file.
The len parameter was changed from unsigned int to size_t in
567c67c6c8.
This fixes crashes in the reference C code, when running checkasm
on aarch64.
Signed-off-by: Martin Storsjö <martin@martin.st>
Terminating the whole checkasm process is not very helpful. This will
report if an illegal instruction occurs while executing a tested
function. This is a common occurrence whilst developping RISC-V
assembler, due to the compatibility between vector configuration and
instruction done at run-time.
For fate-h264_mp4toannexb_ticket5927 and
fate-h264_mp4toannexb_ticket5927_2, they work by accident
previously. The sample file has two 'avc1' entries, and video
samples use the second one. It means packets should be decoded with
new extradata in side data. Before this patch, only extradata was
kept in the output, new extradata has been dropped. The output can
be decoded because the two extradata are almost the same, except
level indication. This patch fixed the issue, and add another
fate test.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
If there is a single group of SPS/PPS before an IDR frame, but no
SPS/PPS after that, we will miss the chance to reset
idr_sps_seen/idr_pps_seen. No SPS/PPS are inserted afterwards.
This patch saves in-band SPS/PPS and insert them before IDR frames
when necessary.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
decorrelate_ls, _rs and _ms are decorrelate[1], [2] and [3] respectively.
The code ended up testing indep ([0]) as twice, ms never, and misnaming
the other two.
Current code tracks min/max pts for each stream separately; then when
the file ends it combines them with last frame's duration to compute the
total duration of each stream; finally it selects the longest of those
durations as the file duration.
This is incorrect - the total file duration is the largest timestamp
difference between any frames, regardless of the stream.
Also change the way the last frame information is reported from decoders
to the muxer - previously it would be just the last frame's duration,
now the end timestamp is sent, which is simpler.
Changes the result of the fate-ffmpeg-streamloop-transcode-av test,
where the timestamps are shifted slightly forward. Note that the
matroska demuxer does not return the first audio packet after seeking
(due to buggy interaction betwen the generic code and the demuxer), so
there is a gap in audio.
The tested functions treat s_m[i] == 0 as a special case. Other than
that, the functions are slightly complicated vector additions.
This actually makes the zero case happen pseudorandomly.
PGMYUV seems to be always limited range. This was a format originally
invented by FFmpeg at a time when YUVJ distinguished limited from full
range YUV, and this codec never appeared to output YUVJ in any
circumstance, so hard-coding limited range preserves the status quo.
The other formats are explicitly documented to be full range RGB/gray
formats. That said, don't tag them yet, due to outstanding bugs w.r.t
grayscale formats and color range handling.
This change in behavior updates a bunch of FATE tests in trivial ways
(added tagging being the only difference).
These are not supported by the drawing functions at all, and were
incorrectly advertised as supported in the past.
Note: This check is added only to separate the logic change from the API
change in the following commit, and will be removed again after it
becomes redundant.
Mapping to ITU-R BS.2051-3 "Sound System J" and ITU-R BS.1196-8 "Channel
Configuration 19".
Signed-off-by: Will Wolcott <wwolcott@netflix.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This layout maps to ITU-R BS.2051-3 "Sound System C" and ITU-R BS.1196-8 "Channel
Configuration 14", and it being the first layout with top layer channels, it's
best to use a different scheme to properly convey the presence and amount of said
channels.
The new name will also be a better fit for the additions in the following commits.
Signed-off-by: James Almer <jamrial@gmail.com>
Fix rendering of int values within a side data element, which was
broken since commit d2d3a83ad9, where the side data element was
correctly marked as a variable fields element. Logic to render a
string variable was implemented already, but it was not implemented
for the int fields path, which was enabled by that commit.
Also, code and schema is changed in order to account for multiple
variable-fields elements - such as side data, contained within the
same parent. Previously it was assumed that a single variable-fields
element was contained within the parent, which was the case for tags,
but is not the case for side-data.
Previously data was rendered as:
<side_data_list>
<side_data side_data_type="CPB properties" max_bitrate="0" min_bitrate="0" avg_bitrate="0" buffer_size="327680" vbv_delay="-1"/>
</side_data_list>
Now as:
<side_data_list>
<side_data type="CPB properties">
<side_datum key="side_data_type" value="CPB properties"/>
<side_datum key="max_bitrate" value="0"/>
<side_datum key="min_bitrate" value="0"/>
<side_datum key="avg_bitrate" value="0"/>
<side_datum key="buffer_size" value="49152"/>
<side_datum key="vbv_delay" value="-1"/>
</side_data>
</side_data_list>
Variable-fields elements are rendered as a containing element wrapping
generic key/values elements, enabling use of strict XML schema.
Fix trac issue:
https://trac.ffmpeg.org/ticket/10613
Always use the functionality of the latter, which makes more sense as it
avoids losing keyframes due to vsync code dropping frames.
Deprecate the 'source_no_drop' value, as it is now redundant.
This is no longer needed as the side data is available for decoders in the
AVCodecContext.
The tests affected reflect the removal of useless CPB and Stereo 3D side
data in packets.
Signed-off-by: James Almer <jamrial@gmail.com>
The changed references for fate-hevc-dv-rpu fate-mov-zombie happen because,
unlike ffmpeg and ffplay, ffprobe never injected packet side data, so the
display matrix side data at the container level is now present in the output
frames.
Signed-off-by: James Almer <jamrial@gmail.com>
Also, avoid spurious end-of-line after side data entries, and improve
rendering of compact output, by adding an indication of the side data
type for each entry.
Also fixes issue:
http://trac.ffmpeg.org/ticket/9266
In this case any timestamps are guessed by compute_pkt_fields() in
libavformat. Since we are decoding the stream, we have more accurate
information from the decoder and do not need any guesses.
Eliminates spurious PTS gaps in a number of FATE tests.
Also avoids dropping the majority of frames in fate-dirac*
We have to write an explicit BlockDuration element (and use
a BlockGroup instead of a SimpleBlock) in case the Track
has a DefaultDuration that is inconsistent with the duration
of the packet.
The matroska-h264-remux test uses a file with coded fields
where the duration of a Block is the duration of a field,
not of a frame, therefore this patch writes said BlockDuration
elements.
(When using a BlockGroup, one has to add ReferenceBlock elements
to distinguish keyframes from non-keyframes. Unfortunately,
the AV1 codec mapping [1] requires us to reference all references
and to really use the real references, which requires a lot of
effort for basically no gain. When BlockGroups are used with AV1,
the created files are most likely invalid, both before and after
this patch, but this patch makes this more likely to happen.)
[1]: https://github.com/ietf-wg-cellar/matroska-specification/blob/master/codec/av1.md
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
They were replaced by TX from libavutil; the tremendous work
to get to this point (both creating TX as well as porting
the users of the components removed in this commit) was
completely performed by Lynne alone.
Removing the subsystems from configure may break some command lines,
because the --disable-fft etc. options are no longer recognized.
Co-authored-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This already avoids unnecessary indirectly included headers
in the arch-specific vf_bwdif_init.c files; it is also in
preparation for splitting the actual functions out of vf_bwdif.c.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
From the documentation of GNU awk [1]:
"In most awk implementations, including gawk, rand() starts generating
numbers from the same starting number, or seed, each time you run awk.45
Thus, a program generates the same results each time you run it. The
numbers are random within one awk run but predictable from run to run.
This is convenient for debugging, but if you want a program to do
different things each time it is used, you must change the seed to a
value that is different in each run. To do this, use srand()."
This commit does exactly this.
[1]: https://www.gnu.org/software/gawk/manual/html_node/Numeric-Functions.html#index-rand_0028_0029-function
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Instead use float one by default for sample rate conversions.
The s16p internal transfer format produces visible and hearable
quantization artifacts.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
for S8 we continue to use S16 as it should have enough precision
Fate is adjusted so bitexactness is maintained between mips/arm/x86
if more tests became bit-inexact on some platform, the same change
can be done to them
The use of higher precision and float intermediates inevitably
leads to more differences between platforms.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It is badly named (should have been -top_field_first, or at least -tff),
underdocumented and underspecified, and (most importantly) entirely
redundant with the setfield filter.
Not every user of idctdsp.h wants to initialize an IDCTDSPContext;
e.g. the proresdsp only uses ff_init_scantable_permutation()
and the IDCT permutation enum; similarly for cavsdsp and wmv2dsp.
Using a forward declaration here avoids an avcodec.h dependency
in the relevant files.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
And remove the AVOID_PROBING flag, given it's the last av1 decoder to be tested
either way.
This fixes a regression introduced in 1652f2492f,
where even if forcing the native av1 decoder, if another decoder was present,
like libdav1d or libaom-av1, they'd be used for probing and some fate tests
would have different results.
Signed-off-by: James Almer <jamrial@gmail.com>
Makes the output of the native decoder consistent with external decoders like
libdav1d with fate-enhanced-flv-av1.
Reviewed-by: Zhao Zhili <quinkblack@foxmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes the test when the non-bitexact MMXEXT versions of
the hpeldsp functions are used.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This makes the test stricter because it is checked that the
MMX registers are not accidentally clobbered.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Only sub_media_pred has an MMXEXT version, so one can use
the version with the stricter check (that checks that
the MMX registers have not been clobbered) for sub_left_predict.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Only the idct_dc and add_residual functions have MMX versions,
so one can use the version with the stricter check (that checks
that the MMX registers have not been clobbered) for all the other
checks.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is the very last user of any lavc transform code.
This also *corrects* wmavoice decoding, as the previous DCT/DST
transforms were incorrect, bringing it closer to Microsoft's
own wmavoice decoder.
Adds a fate test for the jpegxl_anim demuxer, that should allow testing
for true positives and false positives for animated jpegxl files. Note
that two of the test cases are not animated, in order to help sort out
false positives.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
When muxing, the AVStreams' side-data is typically set
by the caller before avformat_write_header();
it is not documented to be else. Yet the Matroska muxer
added an AVStereo3D side data if certain metadata
was present:
Since commit 4d686fb721
(adding support for AVStereo3D stream side-data),
the Matroska muxer checked certain stream tags that
contain Matroska's StereoMode and (if they are present)
converted this value into an AVStereo3D struct that
gets attached to the AVStream (reusing a function from
the demuxer). Afterwards the AVStereo3D side data struct
(whether it has just been added by the muxer or not) gets
parsed and converted back into a Matroska StereoMode.
Besides being an API violation this change broke
StereoMode values without a corresponding AVStereo3D
(namely the anaglyph ones).
This commit fixes this: A StereoMode given via tags
is now used-as-is; if no such tag exists and an AVStereo3D
side data exists, it is converted into the corresponding
StereoMode (if possible). This approach also fixes
handling of the anaglyph ones; the changes to the
matroska-stereo_mode are due to this.
The new STEREOMODE_STEREO3D_MAPPING has been put to
good use for this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
When no output video framerate is specified by the user with -r or can
be inferred from the filtergraph, encoder setup will arbitrarily decide
that the framerate is 25fps. However, making up any framerate value for
VFR encoding is at best unnecessary.
Changes the results of the sub2video tests, where the input timebase is
now used instead of 1/25.
check_func() might return NULL, in which case the function is not to be
benched. Introduced in cc679054c7.
Signed-off-by: Matthias Dressel <code@deadcode.eu>
Signed-off-by: Martin Storsjö <martin@martin.st>
The fate-run.sh shell script exports LC_ALL=C before invoking the
test executables; this is probably done for consistency.
When executing Windows binaries with Wine, it normally handles
UTF-8 command line parameters just fine - but with LC_ALL set to
C, it treats them as plain ASCII.
As the unicode command line parameters wasn't the main thing
being tested here, just convert them to plain ASCII, for
portability. This fixes the test for all test configurations that
use Wine.
Signed-off-by: Martin Storsjö <martin@martin.st>
Writing the duration SimpleTag is special: It's size is
reserved in advance via an EBML Void element (if seekable)
and this reserved space is overwritten when writing the trailer;
it does not use put_ebml_string().
The string to write is created via snprintf on a buffer
of size 20; this buffer is then written via put_ebml_binary()
with a size of 20.
EBML strings need not be zero-terminated; if not, they
are implicitly terminated by the element's length field.
snprintf() always zero-terminates the buffer, i.e.
the last byte can be discarded when using an EBML string.
This patch does this.
The FATE changes are as expected: One byte saved for every
track; the only exception is the matroska-qt-mode test:
An additional byte is saved because an additional byte
could be saved from the enclosing Tags length field.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Do it only for video (the only thing for type for which HDR10+
makes sense).
This effectively reverts changes to several FATE ref-files
made in bda44f0f39.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Provides coverage for the code transforming the ALAC extradata.
Also set creation_time metadata to test this, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Matroska supports orthogonal transformations (both pure rotations
as well as reflections) via its 3D-projection elements, namely
ProjectionPoseYaw (for a horizontal reflection) as well as
ProjectionPoseRoll (for rotations). This commit adds support
for this.
Support for this in the demuxer has been added in
937bb6bbc1 and
the sample used in the matroska-dovi-write-config8 FATE-test
includes a displaymatrix indicating a rotation which is now
properly written and read, thereby providing coverage for
the relevant code in the muxer as well as the demuxer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This fixes the test when running in a cross test setup where the
samples are located at a different path between build host and
temote test target.
Signed-off-by: Martin Storsjö <martin@martin.st>
The contents are full TTML XML documents. TTML writing tests'
results are updated as the streams are now properly identified
as TTML ones.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
This allows this common H.274 SEI to be parsed from both H.264
as well as HEVC, as well as probably from VVC in the future.
Generally attempts to keep the original code as similar as possible.
FATE test refererence changes only change the order of side data
export within a single frame. Nothing else seems to have changed.
This allows this common H.274 SEI to be parsed from both H.264
as well as HEVC, as well as probably from VVC in the future.
Generally attempts to keep the original code as similar as possible.
FATE test refererence changes only change the order of side data
export within a single frame. Nothing else seems to have changed.
The code was blindly assuming that Zbb or V implied Zba. While the
earlier is practically always true, the later broke some QEMU setups,
as V was introduced earlier than Zba.
Read the timebase from FrameData rather than the input stream. This
should fix#10393 and generally be more reliable.
Replace the use of '-1' to indicate demuxing timebase with the string
'demux'. Also allow to request filter timebase with
'-enc_time_base filter'.
Useful for discovering bugs that depend on a specific thread count.
Use like THREADS=randomX for a random thread count from 1 to X, with
X=16 when not specified. Note that the thread count is different for
every test.
Should set "number of frames" to bytes 24-27 of IVF header, not
duration.
It is described by [1], and confirmed by parsing all IVF files in [2].
This commit also updates the md5sum of refs to pass fate-cbs.
[1] Duck IVF - MultimediaWiki
https://wiki.multimedia.cx/index.php/Duck_IVF
[2] webm/vp8-test-vectors
https://chromium.googlesource.com/webm/vp8-test-vectors
Signed-off-by: Jianhui Dai <jianhui.j.dai@intel.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Changes the result of fate-mxf-probe-dv25, where the bitrate is now
exported.
Also changes the result of fate-bsf-dv-error-marker, where the exported
bitrate is now different. Note that the codec layer bitrate does not
match the container bitrate, because container timing is 25fps, while
the DV profile is 50.
Add an optional filter_line3 to the available optimisations.
filter_line3 is equivalent to filter_line, memcpy, filter_line
filter_line shares quite a number of loads and some calculations in
common with its next iteration and testing shows that using aarch64
neon filter_line3s performance is 30% better than two filter_lines
and a memcpy.
Adds a test for vf_bwdif filter_line3 to checkasm
Rounds job start lines down to a multiple of 4. This means that if
filter_line3 exists then filter_line will not sometimes be called
once at the end of a slice depending on thread count. The final slice
may do up to 3 extra lines but filter_edge is faster than filter_line
so it is unlikely to create any noticable thread load variation.
Signed-off-by: John Cox <jc@kynesim.co.uk>
Signed-off-by: Martin Storsjö <martin@martin.st>
No need to generate intermediate files and probe them. We only care to know that the
output of the bsf excludes the frames in question, and a simple checksum is enough.
Signed-off-by: James Almer <jamrial@gmail.com>
Should fix integer overflows, and improve encoding results.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixed-point AAC decoder currently does not produce the same output on
all platforms. Until that is fixed, silence the audio stream using the
volume filter.
Also, actually use the aac_fixed decoder as was the original intent.
The code will currently add a small offset to avoid exact midpoints, but
this can cause inexact results when a float timestamp is exactly
representable as an integer.
Fixes off-by-one in the first frame duration in multiple FATE tests.
Jan Ekström
Marton Balint
sunyuechi
Niklas Haas
Andreas Rheinhardt
rcombs
Connor Worley
Anton Khirnov
Cosmin Stejerean
James Almer
Wu Jianhua
Michael Niedermayer
Martin Storsjö
Marth64
Clément Bœsch
Vignesh Venkatasubramanian via ffmpeg-devel
Thilo Borgmann
Zhao Zhili
Gyan Doshi
Vittorio Giovara
Leo Izen
Paul B Mahol
Rémi Denis-Courmont
Marvin Scholz
Will Wolcott
Stefano Sabatini
Fei Wang
Steven Liu
Steven Liu
Lynne
Matthias Dressel
Jan Ekström
Tobias Rapp
Dai, Jianhui J
John Cox
Tomas Härdin
Romain Beauxis