add tests/ref/fate/filter-hls-append for FATE
add hls-list-append fate use filter make audio data and test hls_flags
append options
Signed-off-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Previously, we used a different list of checks when deciding whether to
write a set of tags at all than we did when deciding whether to write an
individual tag in the set. This resulted in sometimes writing an empty
tag master and seekhead. Now we use mkv_check_tag_name everywhere, so
if a dictionary is entirely composed of tags we skip, we don't write a
tag master at all.
This affected the test file, since "language" was on one list but not
the other, so we were writing an empty tag master there. The test hash
is updated to reflect that change.
Allows testing simple_idct12 correctness/bitexactness, as the sample
was generated using faani as idct.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If this still doesnt give the same results on all platforms then this should be
disabled
Found-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix the demuxer dependencies in some of the tests and remove the vp8 decoder
dependency for the stream copy tests
Reviewed-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This preserves all the information in the codec parameters.
The wavpack ref changes are caused by the fact that now the sample
format is set, so matroskaenc can use it to set the bit depth.
Bug-Id: 945, along with the previous commit
* commit '7ca4fdc046a945949866bb6fc550e3cab1fc8c00':
fate: More fine-grained dependencies for demuxer tests
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '07377fb557909015338b4c215b87b9c73ea577ad':
fate: More fine-grained dependencies for voice codec tests
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '1982d0cc561912d685a0c2dbe58bc19f50bae231':
fate: Add test for MTS2/MSS4
The timestamps differ because we use a more appropriate timebase.
Merged-by: Clément Bœsch <clement@stupeflix.com>
* commit '5b1409c75563b4a3aca113c34d09e3b5442de47f':
fate: Add test for MSS1
Test was already present, see 849e55e58e.
The merge removes the audio decoding present in our version and
simplifies the rules.
Merged-by: Clément Bœsch <clement@stupeflix.com>
* commit '18019f8cb9663dd1032c65aa453eaec18d641905':
FATE: add an H.264 test with unescaped extradata
FATE: add an H.264 test with invalid reference lists
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This tests automatic insertion of the vp9_superframe BSF as well as
ensuring that the colorspace properties in the video header can be
modified when remuxing (-c:v copy).
Also temporary enable the test so we get updated fate failure statistics
Note, this does not work on all platforms, it fails on MIPS
and ml archives indicate it failed on x86 openbsd with some compilers as well
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'b298b36fc008ad94a24929fe770c8189d96bcac4':
fate: Only run SRTP test if SRTP code is enabled
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '30e9ef21cea09fa5e880e979c9f5b39edccbb6f4':
timefilter-test: Only compile timefilter-test if JACK is enabled
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '11843ededacd0157aea642771837557549b5b417':
fate: Add separate target for all indeo3 tests
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'b39ab8549a53e2fc7978ab9db50e5c2ba6a6602d':
fate: Add test for indeo2 with delta frames
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Restore alphabetical order in lists, break overly long lines, do some
prettyprinting, add some explanatory section comments, group parts
together that belong together logically.
The current sample comes from an older version of the codec, which
supports a single output mode, so rename it accordingly.
Add tests for the new pixel formats.
Note some tests need vsync drop to produce exact timestamps, these seem not to
need it. quite likely many more dont need it either, ive not checked beyond finding
one that needs it and the ones which have it removed
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
No new reference samples are needed for this as the file already exists
for testing the bitstream filter
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
- Check if av_display_rotation_get() gets the correct degrees
- Check if av_display_rotation_set() sets the correct matrix
- Check if av_display_matrix_flip() changes correct the matrix
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit also disables the async fate test, because it
used internal APIs in a non-kosher way, which no longer
exists.
* commit '2758cdedfb7ac61f8b5e4861f99218b6fd43491d':
lavf: reorganize URLProtocols
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This options is only used by huffyuv, ffvhuv, jpegls, mjpeg,
mpegvideoenc, png, utvideo.
It is a very codec-specific options, so deprecate the global variant.
Set proper limits to the maximum allowed values, and update utvideoenc
tests to use the new option name.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This option is only used by mpegvideoenc, x264, xavs, and vpx.
It is a very codec-specific option, so deprecate the global variant.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
I/S energy, especially when it comes to phase cancellations,
needs to use signed coefficients as input, yet it was using
abs'd coefficients. That was a slight bug.
The relative error between two encoding strategies is the simple
difference of rate-distortion values, and not the absolute
difference. An absolute measure would allow worsening of the
quantization error as well as improving.
1. Fix sf_idx and band_type addressing to address only the first
subwindow in the group (others could hold garbage values)
2. Don't step on ms_mask when is_mask is set. I/S selection
already sets the ms_mask properly and shouldn't be overridden.
3. Use mid/sid cb/sf when computing coding error, as should be
since those are the cb/sfs that will eventually be set.
4. Fix distortion computation on multi-subwindow groups (was
subtracting the bits terms multiple times)
5. Clear ms_mask when one side uses PNS and the other doesn't.
When using PNS, ms_mask signals correlated noise, which can be
detected just like regular M/S detection, so we don't skip
noise bands, but when only one side uses PNS setting the flag
can confuse some encoders, so avoid that.
* commit 'aebf07075f4244caf591a3af71e5872fe314e87b':
dca: change the core to work with integer coefficients.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Fixes Ticket #5032
The samples in Ticket #5032 is using \r\r\n as line breaks. Since we
already are handling \r, or \n, or \r\n as line breaks, \r\n\n will be
considered as a double line breaks. This is an issue because
ff_subtitles_read_text_chunk() will as a result stop extracting a chunk
after just one line.
So instead of parsing the SRT by "chunks" (which means splitting every
double LB), this new parser is detecting timing lines, and split the
events on this basis. While this sounds safe and simple, it needs to
take into account the event number preceding the timing line while
handling situations such as:
- event number starting at 0 or actually any number instead of 1
- event numbers not being ordered at all
- event number being followed by text garbage (this really happened,
see Ticket #4898)
- event payload containing one or multiple number (a protagonist saying
a count-down, a date or whatever) which could be confused with a
chapter number
- event number being empty (see Ticket #2167)
- all kind of weird line breaks can appear randomly like wild pokémons
- untrustable line breaks (Ticket #5032)
The sample madness.srt tries to sum up most of this into one sample,
ticket5032-rrn.srt is the file containing \r\r\n line breaks. and
empty-events-2167.srt contains empty events.
The DCA core decoder converts integer coefficients read from the
bitstream to floats just after reading them (along with dequantization).
All the other steps of the audio reconstruction are done with floats
which makes the output for the DTS lossless extension (XLL)
actually lossy.
This patch changes the DCA core to work with integer coefficients
until QMF. At this point the integer coefficients are converted to floats.
The coefficients for the LFE channel (lfe_data) are not touched.
This is the first step for the really lossless XLL decoding.
This should fix this test failing on kfreebsd, a regression since
6e5dbe7, which decreased the CMP_TARGET by 1.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
With only 7 coefficients per short window at most the extra precision
makes a difference and seems to reduce crackling and stddev even
further.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This patch does 4 things, all of which interact and thus it
woudln't be possible to commit them separately without causing
either quality regressions or assertion failures.
Fate comparison targets don't all reflect improvements in
quality, yet listening tests show substantially improved quality
and stability.
1. Increase SF range utilization.
The spec requires SF delta values to be constrained within the
range -60..60. The previous code was applying that range to
the whole SF array and not only the deltas of consecutive values,
because doing so requires smarter code: zeroing or otherwise
skipping a band may invalidate lots of SF choices.
This patch implements that logic to allow the coders to utilize
the full dynamic range of scalefactors, increasing quality quite
considerably, and fixing delta-SF-related assertion failures,
since now the limitation is enforced rather than asserted.
2. PNS tweaks
The previous modification makes big improvements in twoloop's
efficiency, and every time that happens PNS logic needs to be
tweaked accordingly to avoid it from stepping all over twoloop's
decisions. This patch includes modifications of the sort.
3. Account for lowpass cutoff during PSY analysis
The closer PSY's allocation is to final allocation the better
the quality is, and given these modifications, twoloop is now
very efficient at avoiding holes. Thus, to compute accurate
thresholds, PSY needs to account for the lowpass applied
implicitly during twoloop (by zeroing high bands).
This patch makes twoloop set the cutoff in psymodel's context
the first time it runs, and makes PSY account for it during
threshold computation, making PE and threshold computations
closer to the final allocation and thus achieving better
subjective quality.
4. Tweaks to RC lambda tracking loop in relation to PNS
Without this tweak some corner cases cause quality regressions.
Basically, lambda needs to react faster to overall bitrate
efficiency changes since now PNS can be quite successful in
enforcing maximum bitrates, when PSY allocates too many bits
to the lower bands, suppressing the signals RC logic uses to
lower lambda in those cases and causing aggressive PNS.
This tweak makes PNS much less aggressive, though it can still
use some further tweaks.
Also update MIPS specializations and adjust fuzz
Also in lavc/mips/aacpsy_mips.h: remove trailing whitespace
"Fast seek" uses linear interpolation to find the position of the
requested seek time. For CBR this is more direct than using the
mp3 TOC and bypassing the TOC avoids problems with TOC precision.
(see https://crbug.com/545914#c13)
For VBR, fast seek is not precise, so continue to prefer the TOC
when available (the lesser of two evils).
Also, some re-ordering of the logic in mp3_seek to simplify and
give usetoc=1 precedence over fastseek flag.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
As noted in a comment, pe.min in the reference encoder
is centered around current pe. The bit reservoir algo
needs pe.min to be a local minimum, because it can only
account for local PE variations. If it's set to a global
minimum as was being done, bit reservoir logic doesn't
work as efficiently.
This patch tries to forget old minimums and converge to
a local minimum without losing the stability of the
previous solution. Listening tests until now suggest this
solves numerous RC issues.
* commit '823fa7004571cb8404ca5785f9fa6e85f0f9f3d3':
fate: Rework sgi tests into a suite and add the missing ones
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
All diferences in unit tests have been acounted for.
* commit '59e8ec0aa8ab174701d01a3bfe96fedd0b7fcead':
movenc: Add an API unit test for fragmenting options/calls
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Contrary to the normal fate tests that run via avconv, this tests
nontrivial call sequences that are only doable via the API
(mainly for different corner cases when using the muxer for
segmenting).
The test muxes fake packet data (with extradata that looks
enough like proper data to make the file be viewable with e.g.
boxdumper) and checks the hash of the produced files. The test also
verifies that fragments produced via different call sequences remain
identical (to avoid e.g. updating the output hashes and suddenly
having fragments that used to be identical suddenly diverging), for
fragments written with frag_discont and/or delay_moov.
Signed-off-by: Martin Storsjö <martin@martin.st>
The CMP variable seems to have been inherited from fate-api-seek which set it to null
the mxf reference needed a change due to c7e14a279f
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Similar to testsrc, but using drawutils and therefore
supporting a lot of pixel formats instead of just rgb24.
This allows using it as input for other tests without
requiring a format conversion.
It is also slightly faster than testsrc for some reason.
This fixes a fate failure after bumping the minor version
Its unknown why this is not needed for the other aac tests,
more investigation needed but for now i dont want to leave
it broken while its investigated
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
There were some errors in the calculation as well as an entire
unnecessary loop to find the gain coefficient. Merge the
two loops.
Thanks to @ubitux for the suggestions and testing.
Michael Niedermayer
Mark Reid
Sasi Inguva
Clément Bœsch
Paul B Mahol
Steven Liu
Burt P
Rodger Combs
Diego Biurrun
Carl Eugen Hoyos
Christophe Gisquet
Jan Sebechlebsky
Steven Robertson
Petru Rares Sincraian
James Almer
Chris Cunningham
Anton Khirnov
Clément Bœsch
Vittorio Giovara
Martin Vignali
Hendrik Leppkes
Muhammad Faiz
Derek Buitenhuis
Ronald S. Bultje
Tobias Rapp
Benjamin Steffes
Claudio Freire
Luca Barbato
Marton Balint
Sasi Inguva
NagaChaitanya Vellanki
Timothy Gu
foo86
Rostislav Pehlivanov
James Darnley
Aman Gupta
Alexandra Hájková
Andreas Cadhalpun
Matthieu Bouron
Simon Thelen
Daniil Cherednik
Martin Storsjö
Nicolas George