Fixes: runtime error: left shift of 32 by 26 places cannot be represented in type 'int'
Fixes: 628/clusterfuzz-testcase-6187747641393152
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The quantization table is stored in the natural order, but when we
access it, we use an index that's in zigzag order, causing us to read
the wrong value. This causes artifacts, especially in areas with
horizontal or vertical edges. The artifacts look a lot like the
DCT ringing artifacts you'd expect to see from a low-bitrate file,
but when comparing to NewTek's own decoder, it's obvious they're not
supposed to be there.
Fix by simply storing the scaled quantization table in zigzag order.
Performance is unchanged.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If the PVQ search picked a place to increment/decrement on the y[]
vector which had no pulse then it would cause a desync since it would
change the sum in the wrong direction. Fix this by not considering
places without pulses as viable.
This makes the PVQ search slightly worse at K < 5 which isn't all that
common. Still, this is a workaround to prevent making broken files until
I can think of a better way of fixing it.
Also add an assertion, which can be removed or moved to assert1/2 once
the PVQ search is stable.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Fixes: runtime error: shift exponent 132 is too large for 32-bit type 'int'
Fixes: 609/clusterfuzz-testcase-4825202619842560
See 11.2.2 IHDR Image header
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If there is progressive input it will disable deinterlacing in cuvid for
all future frames even those interlaced.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
Channel mapping 2 additionally supports a non-diegetic stereo track
appended to the end of a full-order ambisonics signal, such that the
total channel count is either
(n + 1) ^ 2, or
(n + 1) ^ 2 + 2
where n is the ambisonics order
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This marks the first time anyone has written an Opus encoder without
using any libopus code. The aim of the encoder is to prove how far
the format can go by writing the craziest encoder for it.
Right now the encoder's basic, it only supports CBR encoding, however
internally every single feature the CELT layer has is implemented
(except the pitch pre-filter which needs to work well with the rest of
whatever gets implemented). Psychoacoustic and rate control systems are
under development.
The encoder takes in frames of 120 samples and depending on the value of
opus_delay the plan is to use the extra buffered frames as lookahead.
Right now the encoder will pick the nearest largest legal frame size and
won't use the lookahead, but that'll change once there's a
psychoacoustic system.
Even though its a pretty basic encoder its already outperforming
any other native encoder FFmpeg has by a huge amount.
The PVQ search algorithm is faster and more accurate than libopus's
algorithm so the encoder's performance is close to that of libopus
at zero complexity (libopus has more SIMD).
The algorithm might be ported to libopus or other codecs using PVQ in
the future.
The encoder still has a few minor bugs, like desyncs at ultra low
bitrates (below 9kbps with 20ms frames).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>