And remove the unnecessary ffmpeg dependencies while at it.
Reviewed-by: Soft Works <softworkz@hotmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes trac issue #7473.
Removes encoder delay (skip samples) and writes remaining frame samples after EOF to get correct sample count.
Output is now accurate vs players that use Microsoft's codecs (Windows Media Format Runtime).
Tested vs encode>decode WMAv2 with MS's codecs and most sample rate/bit rate/channel/mode combinations in ASF/XWMA.
WMAv1 appears to use the same delay, from FFmpeg samples.
Signed-off-by: bnnm <bananaman255@gmail.com>
subtitles.mak's fate-sub tests utilize a more strict comparator
("rawdiff"), which causes the tests fail in case of white space
differences, such as CRLF vs LF. This in turn causes these
ffprobe-using TTML-in-MP4 tests to fail on non-LF systems such as
Windows or wine.
Includes basic support for both the ISMV ('dfxp') and MP4 ('stpp')
methods. This initial version also foregoes fragmentation support
in case the built-in sample squashing is to be utilized, as this
eases the initial review.
Additionally, add basic tests for both muxing modes in MP4.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
Up until now, the Matroska muxer did not use the dispositions it is
given as-is; instead it by default overrode the disposition of the first
track of a kind (audio, video, subtitles) if no track of this kind has
the default disposition set. And up until recently, it also enforced
by default that no more than one track of each kind be marked as
default.
The rationale for the former is that there are lots of containers which
lack the concept of default streams, so that it is not uncommon for no
stream to be marked as default at all; the rationale for the latter was
that up until recently, it was dubious whether the Matroska specification
allowed more than one default stream for track type (e.g. mkvmerge
disallowed it). It was this point which led to the implementation of
the above mentioned behaviour inspired by mkvmerge.
Yet the Matroska specifications have changed and now explicitly allow
to set more than one track of each type as default, so that the main
reason of not using the dispositions as-is was rendered moot. Therefore
this commit changes the default to pass the disposition through.
The matroska-mpegts-remux FATE-test has been updated to still use the
old "infer" mode so that it is still covered by FATE; the
matroska-zero-length-block test has also been updated to cover
the infer_no_subs mode. The references for lots of other FATE tests
needed to be updated because of a newly added FlagDefault element with
value zero (whereas a FlagDefault with value 1 needn't be coded at all,
as it coincided with the default value of said element).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also adapt some FATE tests to already cover this.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Adds schema validation for ffprobe XML output so that updating the
ffprobe.xsd file upon changes to ffprobe is not forgotten. This was
suggested by Marton Balint in:
http://ffmpeg.org/pipermail/ffmpeg-devel/2021-March/278428.html
The schema FATE test is only run if xmllint command is available.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
After fixing AV_PKT_DATA_SKIP_SAMPLES for reading vorbis packets from ogg,
the actual decoded samples become fewer. Three fate tests are failing:
fate-vorbis-20:
The samples in 6.ogg are not frame aligned. 6.pcm file was generated by
ffmpeg before the fix. After the fix, the decoded pcm file does not match
anymore. Ideally the ref file 6.pcm should be updated but it is probably
not worth it including another copy of the same file, only smaller.
SIZE_TOLERANCE is added for this test case.
fate-webm-dash-chapters:
The original vorbis_chapter_extension_demo.ogg is transmuxed to dash-webm.
The ref file webm-dash-chapters needs to be updated.
fate-vorbis-encode:
This exposes another bug in the vorbis encoder that initial_padding is not
correctly set. It is fixed in the previous patch.
Signed-off-by: Guangyu Sun <gsun@roblox.com>
The twoloop coder is highly loaded with (pseudo-)perceptual metrics,
and the aim of the tests is to piece-wise test each function of the
encoder, for which the 'fast' coder is perfect, since it only decides
on which scalefactors to use, rather than enable or disable encoder
features.
This simply performs a 2nd pass if a LSE is encountered with GRAY8
Fixes: tickets/3933/128.jls
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Deprecated in c29038f304.
The resample filter based upon this library has been removed as well.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This sadly required making changes to the code itself,
due to the same context needing to be reused for both versions.
The lookup table had to be duplicated for both versions.
Notice that the order of the APIC tracks is currently wrong. This is
a superposition of two bugs: (i) Both muxers write the attached
pictures in the order they arrive in the muxer and not in the
stream_index order, leading to attached pictures that are copied being
written earlier because their timestamp is AV_NOPTS_VALUE, whereas the
timestamp of the encoded pictures is 0. (ii) A bug in the id3v2 parsing
code reverses the order of the parsed pictures.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Specifically test that the WebVTT flavour is correctly mapped to
the Matroska/WebM CodecID and back; and test that dispositions
unsupported by WebM are discarded even when they would be supported
by Matroska.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This makes av_read_frame() return packets with proper timestamps.
As a result, seeking now works in combination with streamcopy.
A FATE-test for this has been added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The test sample has to have no file extension, otherwise probing
happens to work, based off file extension alone, and we want to
test the actual probing function.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Enables writing TTML documents or encoded TTML paragraphs as such
documents.
Additionally, a test for the combined TTML encoder and muxer has
been added to validate that the components still work.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
Some FATE tests use files created by other FATE tests as input files;
this mostly affects the seek tests which use files from vsynth_lena as
well as acodec-pcm as input files. In order to make this possible the
temporary files of all the vsynth* and all acodec-pcm tests are kept.
Yet only a fraction of these files are actually used. This commit
changes this to only keep the files that are actually needed for other
tests. This reduces the size of the tests/data/fate folder after a full
FATE run from 2024727441B to 138739312B.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
AVID streams - currently handled by the AVRN decoder - can be (depending
on extradata contents) either MJPEG or raw video. To decode the MJPEG
variant, the AVRN decoder currently instantiates a MJPEG decoder
internally and forwards decoded frames to the caller (possibly after
cropping them).
This is suboptimal, because the AVRN decoder does not forward all the
features of the internal MJPEG decoder, such as direct rendering.
Handling such forwarding in a full and generic manner would be quite
hard, so it is simpler to just handle those streams in the MJPEG decoder
directly.
The AVRN decoder, which now handles only the raw streams, can now be
marked as supporting direct rendering.
This also removes the last remaining internal use of the obsolete
decoding API.
This provides coverage for writing BlockGroups with BlockAdditional
and ReferenceBlock elements. It also tests setting the hearing impaired
disposition (it fits given that this video has no audio so one needs to
be able to read lips to understand anything).
Reviewed-by: Ridley Combs <rcombs@rcombs.me>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The FATE suite already contains a file containing mastering display
and content light level metadata: Meridian-Apple_ProResProxy-HDR10.mxf
This file is used to test both the Matroska muxer and demuxer.
Reviewed-by: Ridley Combs <rcombs@rcombs.me>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The mxf_d10 muxer is very picky regarding the input it accepts:
The only video accepted is MPEG-2 with absolutely constant bitrate,
i.e. all packets need to have exactly the same size; and only a few
bitrates are accepted.
The sample file used did not abide by this: Writing the first packet
(a video packet) errors out and afterwards an audio packet from the
muxing queue has been written. That's all besides metadata (which this
test is about). The FFmpeg cli returned an error, but said error has
been ignored by the md5 test.
This commit changes the test to actually send a compliant stream to the
muxer, so that it does not error out; furthermore, the test is changed
to explicitly check the metadata instead of it only being implicitly
included in the md5 checksum. The compliant stream is created by our
encoder at runtime.
Finally, the test now also covers writing user-specified
product/company/version identification.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Also, test modifying colorspace properties and the default_mode
passthrough which is used here to create a file that has no default
track at all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It furthermore tests the demuxer's handling of chained SeekHeads,
level 1-elements after the Clusters and the muxer's capability of
writing huge TrackNumbers as well as expanding the Cues' length field
by one byte if necessary to fill the reserved space. It also tests
propagation of metadata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
No longer used by anything.
Unfortunately the old FFT_FLOAT/FFT_FIXED_32 is left as-is. It's
simply too much work for code meant to be all removed anyway.
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.
The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.
The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.
Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.
Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.
This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.
MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.
So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.
Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.
This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.
This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.
SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE - 10709590
DROP DSP - 10702872 - diff: -6.56KiB
DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB
DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB
SOFTCODED TABLES:
BASE - 9685096
DROP DSP - 9678378 - diff: -6.56KiB
DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB
DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB
ARM64:
HARDCODED TABLES:
BASE - 14641112
DROP DSP - 14633806 - diff: -7.13KiB
DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB
DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB
SOFTCODED TABLES:
BASE - 13636238
DROP DSP - 13628932 - diff: -7.13KiB
DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB
DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB
x86:
HARDCODED TABLES:
BASE - 12367336
DROP DSP - 12354698 - diff: -12.34KiB
DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB
DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB
SOFTCODED TABLES:
BASE - 11358094
DROP DSP - 11345456 - diff: -12.34KiB
DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB
DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB
PERFORMANCE (10min random s32le):
ARM32 - before - 39.9x - 0m15.046s
ARM32 - after - 28.2x - 0m21.525s
Speed: -30%
ARM64 - before - 36.1x - 0m16.637s
ARM64 - after - 36.0x - 0m16.727s
Speed: -0.5%
x86 - before - 184x - 0m3.277s
x86 - after - 190x - 0m3.187s
Speed: +3%
Do it only when requested with the AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS
flag.
Drop previous code using the long-deprecated AV_FRAME_DATA_QP_TABLE*
API. Temporarily disable fate-filter-pp, fate-filter-pp7,
fate-filter-spp. They will be reenabled once these filters are converted
in following commits.
One of the inputs to the fate test has an rgba pixel format which needs
to be converted to rgb32 (argb on big-endian) for the hqx filter. Because auto
scaling in the fate test is disabled, this needs a separate scale
filter.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
It depends on the muxer generating the timestamps, which is deprecated
and scheduled for removal on next bump.
A bunch of tests change timestamps, because of ffmpeg.c is not
generating them correctly. This should be fixed later.
While the FATE suite contains a sample file for Musepack 8, it did not
use it to test the decoder; it is only used in the mpc8-demux test that
tests the demuxer via streamcopy. Therefore this commit adds an actual
encoder test.
The test uses the framecrc output, because Musepack SV8 is an encoder
that returns multiple frames for a single packet, so that timing
information in the test output is valueable. Output seeking has been
used in order to limit the size of the ref file as well as to test this
codepath for the first time.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
av1dec should no longer attempt to output empty frames if another decoder
was used for probing and it sucessfully set a pix_fmt ever since 05872c67a4,
so we can re-add the AV_CODEC_CAP_AVOID_PROBING cap.
Signed-off-by: James Almer <jamrial@gmail.com>
changes since v1:
- made into fate test
- fixed c90 warnings
- tests more intermediate formats
- tested on BE mips too
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Filters mostly work in native endianness, but they must output
a specified endianness, usually little: that requires a final
conversion for big endian.
I do not know what's the deal with gif-deal: inserting explicitly
the filters that are implicitly inserted result in less frames in
output. Probably a strange problem of duration.
This is utilized by various media ingests to figure out the bit
rate of the content you are pushing towards it, so write it for
video, audio and subtitle tracks in case at least one nonzero value
is available. It is only mentioned for timed metadata sample
descriptions in QTFF, so limit it only to ISOBMFF (MODE_MP4) mode.
Updates the FATE tests which have their results changed due to the
20 extra bytes being written per track.
Explicitly insert the scale or aresample filter where it would
have been inserted by the negotiation.
Re-enable conversions if it cannot be done easily.
If a conversion is needed in a test, we want to know about it.
If the negotiation changes and makes new conversion necessary,
we want to know about it even more.
The dvbsubtest_filter.ts sample is a filtered version of the Videolan
sample database (samples/sub/dvbsub/dvbsubtest.ts) using Project X. It
originates from ticket #8844.
