For hardware cases where we are forced to have a fixed pool of frames
allocated up-front (such as array textures on decoder output), suggest
a possible workaround to the user if an allocation fails because the
pool is exhausted.
Fixes: shift exponent 32 is too large for 32-bit type 'unsigned int'
Fixes: 65909/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VMIX_fuzzer-519459745831321
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Broken in afa471d0ef. It just happened
to work before due to x86inc.asm previously performing XMM spills in
INIT_MMX mode which was more of a bug than an intentional feature.
Based on the AOMedia Film Grain Synthesis 1 (AFGS1) spec:
https://aomediacodec.github.io/afgs1-spec/
The parsing has been changed substantially relative to the AV1 film
grain OBU. In particular:
1. There is the possibility of maintaining multiple independent film
grain parameter sets, and decoders/players are recommended to pick
the one most appropriate for the intended display resolution. This
could also be used to e.g. switch between different grain profiles
without having to re-signal the appropriate coefficients.
2. Supporting this, it's possible to *predict* the grain coefficients
from previously signalled parameter sets, transmitting only the
residual.
3. When not predicting, the parameter sets are now stored as a series of
increments, rather than being directly transmitted.
4. There are several new AFGS1-exclusive fields.
I placed this parser in its own file, rather than h2645_sei.c, since
nothing in the generic AFGS1 film grain payload is specific to T.35, and
to compartmentalize the code base.
Implementation copied wholesale from dav1d, sans SIMD, under permissive
license. This implementation was extensively verified to be bit-exact,
so it serves as a much better starting point than trying to re-engineer
this from scratch for no reason. (I also authored the original
implementation in dav1d, so any "clean room" implementation would end up
looking much the same, anyway)
The notable changes I had to make while adapting this from the dav1d
code-base to the FFmpeg codebase include:
- reordering variable declarations to avoid triggering warnings
- replacing several inline helpers by avutil equivalents
- changing code that accesses frame metadata
- replacing raw plane copying logic by av_image_copy_plane
Apart from this, the implementation is basically unmodified.
Not directly signalled by AV1, but we should still set this accordingly
so that users will know what the original intended video characteristics
and chroma resolution were.
Not directly signalled by AV1, but we should still set this accordingly
so that users will know what the original intended video characteristics
and chroma resolution were.
H.274 specifies that film grain parameters are signalled as intended for
4:4:4 frames, so we always signal this, regardless of the frame's actual
subsampling.
small cleanup for style, redundant semicolons, goto labels,
in FFmpeg, we put goto labels at brace level.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Fixes: out of array access
Fixes: 67021/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4883576579489792
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This will allow users to pass the Android ApplicationContext which is mandatory
to retrieve the ContentResolver responsible to resolve/open Android content URIS.
av_frame_side_data_get() has a const AVFrameSideData * const *sd
parameter; so calling it with an AVFramesSideData **sd like
AVCodecContext.decoded_side_data (or with a AVFramesSideData * const
*sd) is safe, but the conversion is not performed automatically
in C. All users of this function therefore resort to a cast.
This commit changes this: av_frame_side_data_get() is renamed
to av_frame_side_data_get_c(); furthermore, a static inline
wrapper for it name av_frame_side_data_get() is added
that accepts an AVFramesSideData * const * and converts this
to const AVFramesSideData * const * in a Wcast-qual safe way.
This also allows to remove the casts from the current users.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The latter need not be save, because av_log() expects
to get a pointer to an AVClass-enabled structure
and not only a fake object. If this function were actually
be called in the following way:
const AVClass *avcl = avctx->av_class;
handle_mdcv(&avcl, );
the AVClass's item_name would expect to point to an actual
AVCodecContext, potentially leading to a segfault.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This code uses the AVBPrint API for exactly one av_bprintf()
in a scenario in which a good upper bound for the needed
size of the buffer is available (with said upper bound being
much smaller than sizeof(AVBPrint)). So one can simply use
snprintf() instead. This also avoids the (always-false due to
the current size of the internal AVBPrint buffer) check for
whether the AVBPrint is complete.
Furthermore, the old code used AV_BPRINT_SIZE_AUTOMATIC
which implies that the AVBPrint buffer will never be
(re)allocated and yet it used av_bprint_finalize().
This has of course also been removed.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
AVCodecContext extradata should be an AVCDecoderConfigurationRecord
when bitstream format is avcc. Simply concatenating the NALUs output
by x264_encoder_headers does not form a standard
AVCDecoderConfigurationRecord. The following cmd generates broken
file before the patch:
ffmpeg -i foo.mp4 -c:v libx264 -x264-params annexb=0 bar.mp4
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Many users mistakenly think that hwaccel is an instance of a decoder,
and cannot find the corresponding decoder name in the logs. Log
hwaccel name so user know hwaccel has taken effect.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
These were defined in a way compatible with the Vulkan HEVC acceleration, which
expects bitmasks, yet the fields were being overwritting on each loop with the
latest read value.
Signed-off-by: James Almer <jamrial@gmail.com>
The newer of these two are the separate integers for content light
level, introduced in 3952bf3e98c76c31594529a3fe34e056d3e3e2ea ,
with X265_BUILD 75. As we already require X265_BUILD of at least
89, no further conditions are required.
Both of these two structures were first available with X264_BUILD
163, so make relevant functionality conditional on the version
being at least such.
Keep handle_side_data available in all cases as this way X264_init
does not require additional version based conditions within it.
Finally, add a FATE test which verifies that pass-through of the
MDCV/CLL side data is working during encoding.
These two were added in 28e23d7f348c78d49a726c7469f9d4e38edec341
and 3558c1f2e97455e0b89edef31b9a72ab7fa30550 for version 0.9.0 of
SVT-AV1, which is also our minimum requirement right now.
In other words, no additional version limiting conditions seem
to be required.
Additionally, add a FATE test which verifies that pass-through of
the MDCV/CLL side data is working during encoding.
The second part of this condition is intended to check whether the
current quantisation group is in the first CTU column of the current
tile. The issue is that ctb_to_col_bd gives the x-ordinate of the first
column of the current tile *in CTUs*, while xQg gives the x-ordinate of
the quantisation group *in samples*. Rectify this by shifting xQg by
ctb_log2_size to get xQg in CTUs before comparing.
Fixes FFVVC issues #201 and #203.
In MPEG-2 user data, there can be different types of Closed Captions
formats embedded (A53, SCTE-20, or DVD). The current behavior of the
CC extraction code in the MPEG-2 decoder is to not be aware of
multiple formats if multiple exist, therefore allowing one format
to overwrite the other during the extraction process since the CC
extraction shares one output buffer for the normalized bytes.
This causes sources that have two CC formats to produce flawed output.
There exist real-world samples which contain both A53 and SCTE-20 captions
in the same MPEG-2 stream, and that manifest this problem. Example of symptom:
THANK YOU (expected) --> THTHANANK K YOYOUU (actual)
The solution is to pick only the first CC substream observed with valid bytes,
and ignore the other types. Additionally, provide an option for users
to manually "force" a type in the event that this matters for a particular
source.
Signed-off-by: Marth64 <marth64@proxyid.net>
When all cached frames are drained, the output mfxSyncPoint pointer is
NULL and MFX_ERR_MORE_DATA is returned, hence needn't print warning for
this expected behavior, otherwise the user might think the output from
qsv decoders are wrong.
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
ff_aac_coder_init_mips() modifies a static const structure of
function pointers. This will crash if the binary uses relro
and is a data race in any case.
Furthermore it points to a maintainability issue: The
AACCoefficientsEncoder structures have been constified
in commit fd9212f2ed,
a Libav commit merged in 318778de9e.
Libav did not have the MIPS-specific AAC code and so this was
fine for them; yet FFmpeg had them, but this was not recognized.
Commit 75a099fc73 points to another
maintainability issue: Contrary to ordinary DSP code, this code
here is way more complex and needs to be constantly kept in sync
with the ordinary code which it mimicks and replaces. Said commit
is the only commit actually changing aaccoder.c in the last few
years and the same change has not been performed for the MIPS
clone; before that, it even happened several times that the mips
code was broken due to changes of the generic code (see commits
97437bd17a and
de262d018d or
860dbe0275 or
933309a6ca or
b65ffa316e). This might even lead
to scenarios where someone changing non-dsp aacenc code would
have to modify mips inline asm in order to keep them in sync.
This is obviously a significant burden (if the AAC encoder were
actively developed).
Finally, the code does not even compile here due to errors like
"Error: float register should be even, was 1".
Reviewed-by: Lynne <dev@lynne.ee>
Reviewed-by: Jean-Baptiste Kempf <jb@videolan.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Mark Thompson
Michael Niedermayer
Henrik Gramner
Lynne
Niklas Haas
Jun Zhao
Matthieu Bouron
Andreas Rheinhardt
Zhao Zhili
James Almer
Jan Ekström
Frank Plowman
Marth64
Fei Wang
Haihao Xiang