Use the next I/P/B or start code as the end of current frame.
Before the patch, extension start code, user data start code,
sequence end code and so on are treated as the start of next
frame.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Since this is an external encoder not under our control, we cannot test
the encoded output exactly as is done for internal encoders. We can
still test however that the output is decodable and produces the
expected number of frames with expected dimensions, pixel formats, and
timestamps.
Adds JPEG 2000 decoder tests using materials from the conformance suite specified in
Rec. ITU-T T.803 | ISO/IEC 15444-4.
The test materials are available at https://gitlab.com/wg1/htj2k-codestreams
Signed-off-by: Pierre-Anthony Lemieux <pal@palemieux.com>
When no packet dts values are available from the container, video
decoding code will currently use its own guessed values, which will then
be propagated to pkt_dts on decoded frames and used as pts in certain
cases. This is inaccurate, fragile, and unnecessarily convoluted.
Simply removing this allows the extrapolation code introduced in the
previous commit to do a better job.
Changes the results of numerous h264 and hevc FATE tests, where no
spurious timestamp gaps are generated anymore. Several tests no longer
need -vsync passthrough.
When no timestamps are available from the container, the video decoding
code will currently use fake dts values - generated in
process_input_packet() based on a combination of information from the
decoder and the parser (obtained via the demuxer) - to generate
timestamps during decoder flushing. This is fragile, hard to follow, and
unnecessarily convoluted, since more reliable information can be
obtained directly from post-decoding values.
The new code keeps track of the last decoded frame pts and estimates its
duration based on a number of heuristics. Timestamps generated when both
pts and pkt_dts are missing are then simple pts+duration of the last frame.
The heuristics are somewhat complicated by the fact that lavf insists on
making up packet timestamps based on its highly incomplete information.
That should be removed in the future, allowing to further simplify this
code.
The results of the following tests change:
* h264-3386 now requires -fps_mode passthrough to avoid dropping frames
at the end; this is a pathology of the interaction of the new and old
code, and the fact that the sample switches from field to frame coding
in the last packet, and will be fixed in following commits
* hevc-conformance-DELTAQP_A_BRCM_4 stops inventing an arbitrary
timestamp gap at the end
* hevc-small422chroma - the single frame output by this test now has a
timestamp of 0, rather than an arbitrary 7
Timestamps in two FATE H.264 conformance tests now start at 1 instead
of 0, which also happens in some other H.264 tests before this commit
and so is not a big issue.
Conversely, timestamps in some HEVC conformance tests start from a
smaller value now.
Ideally this should be addressed later in a more general way.
h264-conformance-frext-frext2_panasonic_b no longer requires -vsync
passthrough.
The previous commit allowed turning on error correction for interlaced
samples. Turning it off amounts to a no-op for FATE. These samples
should be tested with EC1-3 (guess_mvs/deblock/favor_inter)
additionally.
Signed-off-by: J. Dekker <jdek@itanimul.li>
According to MXF specs the Stored Rectangle corresponds to the data which is
passed to the compressor and received from the decompressor, so they should
contain the width / height extended to the macroblock boundary.
In practice however width and height values rounded to the upper 16 multiples
are only seen when muxing MPEG formats. Therefore this patch changes stored
width and height values to unrounded for all non-MPEG formats, even macroblock
based ones.
For DNXHD the specs (ST 2019-4) explicitly indicates to use 1080 for 1088p.
For ProRes the specs (RDD 44) only refer to to ST 377-1 without precision but
no known commercial implementations are using rounded values.
DV is not using 16x16 macroblocks, so 16 rounding makes no sense.
The patch also fixes Sampled Width / Display Width to use unrounded values.
Signed-off-by: Marton Balint <cus@passwd.hu>
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>
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).
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 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.
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>
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>
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>
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>
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>
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>
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.
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>
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>
