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.
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.
Currently those are set in different ways depending on whether the
stream is decoded or not, using some values from the decoder if it is.
This is wrong, because there may be arbitrary amount of delay between
input packets and output frames (depending e.g. on the thread count when
frame threading is used).
Always use the path that was previously used only for streamcopy. This
should not cause any issues, because these values are now used only for
streamcopy and discontinuity handling.
This change will allow to decouple discontinuity processing from
decoding and move it to ffmpeg_demux. It also makes the code simpler.
Changes output in fate-cover-art-aiff-id3v2-remux and
fate-cover-art-mp3-id3v2-remux, where attached pictures are now written
in the correct order. This happens because InputStream.dts is no longer
reset to AV_NOPTS_VALUE after decoding, so streamcopy actually sees
valid dts values.
Stop using InputStream.dts for generating missing timestamps for decoded
frames, because it contains pre-decoding timestamps and there may be
arbitrary amount of delay between input packets and output frames (e.g.
dependent on the thread count when frame threading is used). It is also
in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary
rounding issues.
New code maintains a timebase that is the inverse of the LCM of all the
samplerates seen so far, and thus can accurately represent every audio
sample. This timebase is used to generate missing timestamps after
decoding.
Changes the result of the following FATE tests
* pcm_dvd-16-5.1-96000
* lavf-smjpeg
* adpcm-ima-smjpeg
In all of these the timestamps now better correspond to actual frame
durations.
Previously they would only be used with trivial filtergraphs, because
filters did not handle frame durations. That is no longer true - most
filters process frame durations properly (there may still be some that
don't - this change will help finding and fixing them).
Improves output video frame durations in a number of FATE tests.
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 enable lcms2, the fate-png-icc-parse will get error bellow.
TEST png-icc-parse
This because updated how PNG handles colors (no
longer using mastering display metadata for that).
Reviewed-by: Leo Izen <leo.izen@gmail.com>
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
Remove now-obsolete code setting packet durations pre-muxing for CFR
encoded video.
Changes output in the following FATE tests:
* numerous adpcm tests
* ffmpeg-filter_complex_audio
* lavf-asf
* lavf-mkv
* lavf-mkv_attachment
* matroska-encoding-delay
All of these change due to the fact that the output duration is now
the actual input data duration and does not include padding added by
the encoder.
* apng-osample: less wrong packet durations are now passed to the muxer.
They are not entirely correct, because the first frame duration should
be 3 rather than 2. This is caused by the vsync code and should be
addressed later, but this change is a step in the right direction.
* tscc2-mov: last output frame has a duration of 11 rather than 1 - this
corresponds to the duration actually returned by the demuxer.
* film-cvid: video frame durations are now 2 rather than 1 - this
corresponds to durations actually returned by the demuxer and matches
the timestamps.
* mpeg2-ticket6677: durations of some video frames are now 2 rather than
1 - this matches the timestamps.
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
Changes the result of the h264_redundant_pps-mov test, where the output
timebase is now 1001/24000 instead of 1/24. This is more correct, as the
source file actually is 23.98fps.
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.
For audio AVFrames, nb_samples is typically more trustworthy than
duration. Since sync queues look at durations, make sure they match the
sample count.
The last audio frame in the fate-shortest test is now gone. This is more
correct, since it outlasts the last video frame.
These fields are ad-hoc and will be deprecated. Use the recently-added
AV_CODEC_FLAG_COPY_OPAQUE to pass arbitrary user data from packets to
frames.
Changes the result of the flcl1905 test, which uses ffprobe to decode
wmav2 with multiple frames per packet. Such packets are handled
internally by calling the decoder's decode callback multiple times,
offsetting the internal packet's data pointer and decreasing its size
after each call. The output pkt_size value before this commit is then
the remaining internal packet size at the time of each internal decode
call.
After this commit, output pkt_size is simply the size of the full packet
submitted by the caller to the decoder. This is more correct, since
internal packets are never seen by the caller and should have no
observable outside effects.
ISOBMFF (14496-12) made this field ('channelcount') in the
AudioSampleEntry structure non-template¹ somewhere before the
release of the 2022 edition. As for ETSI TS 126 244 AKA 3GPP
file format (V16.1.0, 2020-10), it does not seem contain any
references limiting the channelcount entry in AudioSampleEntry
or in its own definition of EVSSampleEntry.
fate-mov-mp4-chapters test had to be adjusted as it output a
mono vorbis stream, which would now be properly marked as such
in the container.
1: As per 14496-12:
Fields shown as “template” in the box descriptions are fields
which are coded with a default value unless a derived
specification defines their use and permits writers to use
other values than the default.
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>
The cHRM chunk is descriptive. That is, it describes the primaries that should
be used to interpret the pixel data in the PNG file. This is notably different
from Mastering Display Metadata, which describes which subset of the presented
gamut is relevant. MDM describes a gamut and says colors outside the gamut are
not required to be preserved, but it does not actually describe the gamut that
the pixel data from the frame resides in. Thus, to decode a cHRM chunk present
in a PNG file to Mastering Display Metadata is incorrect.
This commit changes this behavior so the cHRM chunk, if present, is decoded to
color metadata. For example, if the cHRM chunk describes BT.709 primaries, the
resulting AVFrame will be tagged with AVCOL_PRI_BT709, as a description of its
pixel data. To do this, it utilizes libavutil/csp.h, which exposes a funcction
av_csp_primaries_id_from_desc, to detect which enum value accurately describes
the white point and primaries represented by the cHRM chunk.
This commit also changes pngenc.c to utilize the libavuitl/csp.h API, since it
previously duplicated code contained in that API. Instead, taking advantage of
the API that exists makes more sense. pngenc.c does properly utilize the color
tags rather than incorrectly using MDM, so that required no change.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
segment_time and segment_times are defined as duration specifications, not
timestamps, so calculation of segment duration must account for initial
timestamp. Fixed.
FATE ref for segment-mp4-to-ts changed on account of avoiding premature
segment cut at the end of the first segment.
Defined by H.274, this SEI message is utilized by iPhones to save
the nominal ambient viewing environment for the display of recorded
HDR content. The contents of the message are exposed to API users
as AVFrame side data containing AVAmbientViewingEnvironment.
As the DV RPU test sample is from an iPhone and includes Ambient
Viewing Environment SEI messages, its test result gets updated.
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).
Currently, in case of equality on the first color channel, the order of
the ref colors is defined by the hashing function. This commit makes the
sorting deterministic and improve the hierarchical ordering.
This filter, when used in the "pad" mode, currently makes the
distinction between limited and full range solely by testing for YUVJ
pixel formats at link setup time. This is deprecated and should be
improved to perform the detection based on the per-frame metadata.
In order to make this distinction based on color range metadata, which
is only known at the time of filtering frames, for simplicity, we simply
allocate two copies of the "black" frame - one for limited range and the
other for full range metadata. This could be done more dynamically (e.g.
as-needed or simply by blitting the appropriate pixel value directly),
but this change is relatively simple and preserves the structure of the
existing code.
This commit actually fixes a bug in FATE - the new output is correct for
the first time. The previous md5 ref was of a frame that incorrectly
combined full-range pixel data with limited-range black fields. The
corresponding result has been updated.
Signed-off-by: Niklas Haas <git@haasn.dev>
The idea behind last_pkt_props was to store the properties of the last packet
fed to the decoder. Any sort of queueing required by CODEC_CAP_DELAY decoders
that consume several packets before they start outputting frames should be done
by the decoders in question. An example of this is libdav1d.
This is required for the following commits that will fix last_pkt_props in
frame threading scenarios, as well as maintain its contents during flush.
This revers commit 022a12b306.
Signed-off-by: James Almer <jamrial@gmail.com>
The Encoding field (and the \fe tag) allows to limit font selection to
only those fonts declaring support for the specified codepage in their
OS/2's table "Code Page Character Range" field.
Particularly, Encoding=0 means only font's declaring support for "ANSI",
or rather "Latin (Western European)", are allowed to be selected.
Specifying Encoding=1 allows all fonts to be considered.
We do not want to limit font selection, so specify Encoding=1.
NB: at the time of writing libass only partially supports this field,
thus hiding the issue in any libass-based renderer. A VSFilter-based
DirectShow filter or XySubFilter will reveal the issue when a font not
declaring support for latin characters is specified in a style.
Colour values used in ASS files without a "YCbCr Matrix" header set to
"None" are subject to colour mangling, due to how ASS was historically
conceived. A more in-depth description can be found in the documetation
inside libass' public ass_types.h header. The important part is, if this
header is not set to "None", the final output colours can deviate from
the literal value specified in the file. When converting from non-ASS
formats we do not want any colour shift to happen, so let's set the
appropiate header.
NB: ffmpeg's subtitle filter, does not follow libass' documentation
regarding colour mangling, thus hiding the bug. Anything based on
VSFilter, XySubFilter or e.g. mpv do and might show the issue.
(Of course native ASS subs, which _do_ rely on colour mangling won't
work properly with the subtitle filter, but this can be fixed another
time)
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>
floating point uses a slightly different predictor technique describe here
http://chriscox.org/TIFFTN3d1.pdf
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This patch replaces the transform used in AAC with lavu/tx and removes
the limitation on only being able to decode 960-sample files
with the float decoder.
This commit also removes a whole bunch of unnecessary and slow
lifting steps the decoder did to compensate for the poor accuracy
of the old integer transformation code.
Overall float decoder speedup on Zen 3 for 64kbps: 32%
Fixes ticket #128.
The SVQ1 interframe mean VLC symbols -128 and 128 are incorrectly swapped
in our SVQ1 implementation, resulting in visible artifacts for some videos.
This patch unswaps the order of these two symbols.
The most noticable example of the artiacts caused by this error can be observed in
https://trac.ffmpeg.org/attachment/ticket/128/svq1_set.7z '352_288_k_50.mov'.
The artifacts are not observed when using the reference decoder
(QuickTime 7.7.9 x86 binary).
As a result of this patch, the reference data for the fate-svq1 test
($SAMPLES/svq1/marymary-shackles.mov) must be modified. For this file, our
decoder output is now bitwise identical to the reference decoder. I have
tested patch with various other samples and they are all now bitwise identical.
The data in SGI images is stored planar, so exporting
it via planar pixel formats is natural.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This check is intended to be avoid buffer overflows,
yet there are four problems with it:
1. It has an in-built off-by-one error: len == out_end - out
is perfectly fine and nothing to worry about.
This off-by-one error led to the pixel in the lower-right corner
not being set properly for the back frame of the sample from
the rl2 FATE-test. This pixel is copied to every frame which
is the reason for the update to the reference file of said test.
With this patch, the output of the decoder matches the output
as captured from the reference decoder* (apart from the fact
that said reference somehow lacks the top part of the frame
(copied over from the background frame)).
2. Given that the stride of the buffer may be different
from the width of the video (despite one pixel taking one byte),
there is a second check lateron making the first check redundant
(if one returns immediately; a simple break at the second check
is not sufficient, because it only exits the inner loop).
3. The check is based around the assumption of the stride being
positive (it has this in common with the other check which
will be fixed in a future commit).
4. Even after fixing the off-by-one error, the check in
question is still triggered by all the non-background frames
in the FATE sample as well as by A1100100.RL2. In all these
cases, they use len == 255 and val == 128. For videos with
background frame this just means "copy from the background
frame", which would be done anyway lateron.* Yet for videos
without it copying it is necessary to avoid leaving
uninitialized parts in the video.
*: Available in https://samples.mplayerhq.hu/game-formats/voyeur-rl2/
**: Due to this, the code that copies the rest from the
back frame is no longer executed for any of the samples
available on the sample server. Given that these are only
the files from the demo version of this game, I don't know
whether this code is executed for any file in existence or not.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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>
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>
Do this by setting AVCodecInternal.pad_samples.
This prevents reading into the frame's padding and writing
into the packet's padding.
This actually happened in our FATE tests (where the number of samples
is 2 mod 4), which therefore needed to be updated.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
APTX decodes four bytes of input to four stereo samples; APTX HD
does the same with six bytes of input. So it can be easily supported
in av_get_audio_frame_duration().
This fixes invalid durations and (derived) timestamps of demuxed
APTX HD packets and therefore fixed the timestamp in the aptx-hd
FATE test.
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>
Since introducing the various packed formats used by VAAPI (and p012),
we've noticed that there's actually a gap in how
av_find_best_pix_fmt_of_2 works. It doesn't actually assign any value
to having the same bit depth as the source format, when comparing
against formats with a higher bit depth. This usually doesn't matter,
because av_get_padded_bits_per_pixel() will account for it.
However, as many of these formats use padding internally, we find that
av_get_padded_bits_per_pixel() actually returns the same value for the
10 bit, 12 bit, 16 bit flavours, etc. In these tied situations, we end
up just picking the first of the two provided formats, even if the
second one should be preferred because it matches the actual bit depth.
This bug already existed if you tried to compare yuv420p10 against p016
and p010, for example, but it simply hadn't come up before so we never
noticed.
But now, we actually got a situation in the VAAPI VP9 decoder where it
offers both p010 and p012 because Profile 3 could be either depth and
ends up picking p012 for 10 bit content due to the ordering of the
testing.
In addition, in the process of testing the fix, I realised we have the
same gap when it comes to chroma subsampling - we do not favour a
format that has exactly the same subsampling vs one with less
subsampling when all else is equal.
To fix this, I'm introducing a small score penalty if the bit depth or
subsampling doesn't exactly match the source format. This will break
the tie in favour of the format with the exact match, but not offset
any of the other scoring penalties we already have.
I have added a set of tests around these formats which will fail
without this fix.
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>
This duration is equal to the longest duration in all track's tkhd atoms, which
may be comprised of the sum of all edit lists in each track. Empty edit lists
in tracks represent start_time, and the actual media duration is stored in the
mdhd atom.
This change lets the generic demux code derive the longest track duration taken
from mdhd atoms, so the correct duration and start_time combination will be
reported.
Should fix ticket #9775.
Reviewed-by: zhilizhao(赵志立) <quinkblack@foxmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The field is not specific to Opus.
The mp2fixed encoder signals initial_padding and is used
by both the matroska-encoding-delay test as well as
the lavf-mkv tests which necessitated several FATE ref changes.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Matroska generally requires timestamps to be nonnegative, but
there is an exception: Data that corresponds to encoder delay
and is not supposed to be output anyway can have a negative
timestamp. This is achieved by using the CodecDelay header
field: The demuxer has to subtract this value from the raw
(nonnegative) timestamps of the corresponding track.
Therefore the muxer has to add this value first to write
this raw timestamp.
Support for writing CodecDelay has been added in FFmpeg commit
d92b1b1bab and in Libav commit
a1aa37dd0b. The former simply
wrote the header field and did not apply any timestamp offsets,
leading to desynchronisation (if one uses multiple tracks).
The latter applied it at two places, but not at the one where
it actually matters, namely in mkv_write_block(), leading to
the same desynchronisation as with the former commit. It furthermore
used the wrong stream timebase to convert the delay to the
stream's timebase, as the conversion used the timebase from
before avpriv_set_pts_info().
When the latter was merged in 82e4f39883,
it was only done in a deactivated state that still did not
offset the timestamps when muxing due to "assertion failures
and av sync errors". a1aa37dd0b
made it definitely more likely to run into assertion failures
(namely if the relative block timestamp doesn't fit into an int16_t).
Yet all of the above issues have been fixed (in commits
962d631573,
5d3953a5dc and
4ebeab15b0. This commit therefore
enables applying CodecDelay, fixing ticket #7182.
There is just one slight regression from this: If one has input
with encoder delay where the first timestamp is negative, but
the pts of the part of the data that is actually intended to be
output is nonnegative, then the timestamps will currently by default
be shifted to make them nonnegative before they reach the muxer;
the muxer will then ensure that the shifted timestamps are retained.
Before this commit, the muxer did not ensure this; instead the
timestamps that the demuxer will output were shifted and
if the first timestamp of the actually intended output was zero
before shifting, then this unintentional shift just cancels
the shift performed before the packet reached the muxer.
(But notice that this only applies if all the tracks use the same
CodecDelay, or the relative sync between tracks will be impaired.)
This happens in the matroska-opus-remux and matroska-ogg-opus-remux
FATE tests. Future commits will forward the information that
the Matroska muxer has a limited capability to handle negative
timestamps so that the shifting in libavformat can take advantage
of it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is possible for the trailing padding to be zero, namely
e.g. if the AV_PKT_DATA_SKIP_SAMPLES side data is used
for leading padding. Matroska supports this (use a negative
DiscardPadding), but players do not; at least Firefox refuses
to play such a file. So for now only write DiscardPadding
if it is trailing padding and nonzero.
The fate-matroska-ogg-opus-remux was affected by this.
(I wish CodecDelay would not exist and DiscardPadding would
be used to instead trim the codec delay away (with the Block
timestamp corresponding to the time at which the actually
output audio is output).)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
These are the formats we want/need to use when dealing with the Intel
VAAPI decoder for 12bit 4:2:0, 12bit 4:2:2, 10bit 4:4:4 and 12bit 4:4:4
respectively.
As with the already supported Y210 and YUVX (XVUY) formats, they are
based on formats Microsoft picked as their preferred 4:2:2 and 4:4:4
video formats, and Intel ran with it.
P12 and Y212 are simply an extension of 10 bit formats to say 12 bits
will be used, with 4 unused bits instead of 6.
XV30, and XV36, as exotic as they sound, are variants of Y410 and Y412
where the alpha channel is left formally undefined. We prefer these
over the alpha versions because the hardware cannot actually do
anything with the alpha channel and respecting it is just overhead.
Y412/XV46 is a normal looking packed 4 channel format where each
channel is 16bits wide but only the 12msb are used (like P012).
Y410/XV30 packs three 10bit channels in 32bits with 2bits of alpha,
like A/X2RGB10 style formats. This annoying layout forced me to define
the BE version as a bitstream format. It seems like our pixdesc
infrastructure can handle the LE version being byte-defined, but not
when it's reversed. If there's a better way to handle this, please
let me know. Our existing X2 formats all have the 2 bits at the MSB
end, but this format places them at the LSB end and that seems to be
the root of the problem.
As we already have support for VUYA, I figured I should do the small
amount of work to support VUYX as well. That means a little refactoring
to share code.
This is the alphaless version of VUYA that I introduced recently. After
further discussion and noting that the Intel vaapi driver explicitly
lists XYUV as a support format for encoding and decoding 8bit 444
content, we decided to switch our usage and avoid the overhead of
having a declared alpha channel around.
Note that I am not removing VUYA, as this turned out to have another
use, which was to replace the need for v408enc/dec when dealing with
the format.
The vaapi switching will happen in the next change
IEEE-754 differentiates two different kind of NaNs.
Quiet and Signaling ones. They are differentiated by the MSB of the
mantissa.
For whatever reason, actual hardware conversion of half to single always
sets the signaling bit to 1 if the mantissa is != 0, and to 0 if it's 0.
So our code has to follow suite or fate-testing hardware float16 will be
impossible.
Up until now, ff_wmv2_decode_secondary_picture_header() only
set the mb_type array for non I-pictures, so that the decoding
process uses the earlier values of this array; this affects
the output of the wmv8-x8intra FATE-test (which this patch
therefore updates). These earlier values were set when decoding
earlier frames or when the buffer was initially zero-allocated.
A consequence of this is that the output of this test would be
random if ff_find_unused_picture() would select the unused picture
to return at random. Furthermore decoding from a keyframe onwards
depends upon the earlier state of the decoder.
This patch therefore zeroes said array when decoding an I picture.
(It is not claimed that zero is the right value to fill the array with.
I just don't know.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The "AYUV" format is defined by Microsoft as their preferred format for
4:4:4 content, and so it is the format used by Intel VAAPI and QSV.
As Microsoft like to define their byte ordering in little-endian
fashion, the memory order is reversed, and so our pix_fmt, which
follows memory order, has a reversed name (VUYA).
Firstly, the timestamps generated from framerate are inaccurate for
variable framerate mode.
Secondly, the timestamps always start from zero, while pts/dts can
start from nonzero. FLV demuxer rejects such index with message:
"Found invalid index entries, clearing the index".
Necessitated by 6ca43a9675
and 425b309fa4.
Reviewed-by: Martin Storsjö <martin@martin.st>
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>
Same issues apply to it as to -shortest.
Changes the results of the following tests:
- matroska-flac-extradata-update
The test reencodes two input FLAC streams into three output FLAC
streams. The last output stream is limited to 8 frames. The current
code results in the first two output streams having 12 frames, after
this commit all three streams have 8 frames and are the same length.
This new result is better, since it is predictable.
- mkv-1242
The test streamcopies one video and one audio stream, video is limited
to 11 frames. The new result shortens the audio stream so that it is
not longer than the video.
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.
The amount of padding samples reported by containers take into account the
extended samplerate in HE-AAC.
Fixes ticket #9671.
Signed-off-by: James Almer <jamrial@gmail.com>
wrapped_avframe_decode() uses an AVFrame as dst in av_frame_move_ref()
after having called ff_decode_frame_props() to attach side-date
to this very frame. This leaks all the side-data and metadata
that ff_decode_frame_props() has attached.
This happens in various fate-filter-metadata tests since
6ca43a9675.
These particular leaks (which affect metadata-only)
could be fixed by not adding metadata side-data to AVPackets
in libavdevice if they are also available from the AVFrames.
Yet this would break users that extract the metadata from
AVPackets.
The changes to FATE happen because of the way av_dict_set()
works when it overwrites an already existing entry:
It overwrites the entry to be overwritten with the last entry
and adds the new entry at the end. The end result is that
the first entry of the dict is the second-to-last-entry of
the original dict, the last entry of the dict is the last
entry of the old dict and the first count - 2 entries
of the original dict are at positions 1..count - 2 in their
original order.
Reviewed-by: Timo Rothenpieler <timo@rothenpieler.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This avoids an extra copy of potentially quite big video frames.
Instead of copying the entire frames data into a rawvideo packet it
packs the frame into a wrapped avframe packet and passes it through
as-is.
Unfortunately, wrapped avframes are set up to be video frames, so the
audio frames continue to be copied.
Additionally, this enabled passing through video frames that previously
were impossible to process, like hardware frames or other special
formats that couldn't be packed into a rawvideo packet.
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>
Up until now, updating extradata was very ad-hoc: The amount of
space reserved for extradata was not recorded when writing the
header; instead the AAC code simply presumed that it was enough.
This commit changes this by recording how much space is available.
This brings with it that the code for writing of and reserving space
for the CodecPrivate and for updating it diverges. They are therefore
split; this allows to put other common tasks like seeking to
right offset as well as writing padding (in case the new extradata did
not fill the whole reserved space) to this common function.
The code for filling up the reserved space is smarter than the code
it replaces; therefore it is no longer necessary to reserve more
than necessary just to be sure that one can add an EBML Void element
(whose minimum size is two) lateron. This is the reason for the change
to the aac-autobsf-adtstoasc test.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is possible by using a dynamic buffer to write them;
said dynamic buffer is (re)used and reset as appropriate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
- Introduce ff_draw_init2, which takes explicit colorspace and range
args
- Use lavu/csp and lavfi/colorspace for conversion, rather than the
lavu/colorspace.h macros
- Use the passed-in colorspace when performing RGB->YUV conversions
The upshot of this is:
- Support for YUV spaces other than BT601
- Better rounding for all conversions
- Particular rounding improvements in >8-bit formats, which previously
used simple left-shifts
- Support for limited-range RGB
- Support for full-range YUV in non-J pixfmts
Due to the rounding improvements, this results in a large number of
minor changes to FATE tests.
Signed-off-by: rcombs <rcombs@rcombs.me>
Fixes CID1396405
MSE and PSNR is slightly improved, and some noticable corruptions disappear as
well.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
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>
Since every DLL can use an individual CRT on Windows, having
an exported function that opens a FILE* won't work if that
FILE* is going to be used from a different DLL (or from user
application code).
Internally within the libraries, the issue can be worked around
by duplicating the function in all libraries (this already happened
implicitly because the function resided in file_open.c) and renaming
the function to ff_fopen_utf8 (so that it doesn't end up exported from
the DLLs) and duplicating it in all libraries that use it.
This makes the avpriv_fopen_utf8 / ff_fopen_utf8 function work in
the exact same way as the existing avpriv_open / ff_open, with the
same setup as introduced in e743e7ae6e.
That mechanism doesn't work for external users, thus deprecate the
existing function.
Signed-off-by: Martin Storsjö <martin@martin.st>
Do this by making this test a transcode test.
Also fix the test requirements and don't add this test to FATE_AFILTER;
instead use a new variable and a new target for flvenc-tests.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Accidentally resurrected in fc49f22c3b
and 7711f19eda,
forgotten in 6ebc71847e and
1a6a088f7c or never needed
(filter-aemphasis).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Each of the intermediately generated lena-*.fits files is only used
for exactly one test; so it could be deleted right after the test.
Switching to a transcode test (which is also more natural) achieves
this. It also adds checksums of the intermediate files to the ref-file.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is mostly straightforward. The major complication is that, as a
result of the 16-bit chunk size limitation, ICC profiles may need to be
split up into multiple chunks.
We also need to make sure to allocate enough extra space in the packet
to fit the ICC profile, so modify both mpegvideo_enc.c and ljpegenc.c to
take into account this extra overhead, failing cleanly if necessary.
Also add a FATE transcode test to ensure that the ICC profile gets
written (and read) correctly. Note that this ICC profile is smaller than
64 kB, so this doesn't test the APP2 chunk re-arranging code at all.
Signed-off-by: Niklas Haas <git@haasn.dev>
