When 'out' is an AVFrame that does not have buffers preallocated,
swr_convert_frame tries to allocate buffers of the right size. However
in calculating this size it failed to check for whether 'in' is NULL
(requesting that swr's internal buffers are to be flushed).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
use fltp when doing s32 -> s32 resampling
because s32p has no simd optimization
benchmark:
old 17.913s
new 7.584s (use fma3)
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
when set_compensation is called with zero sample_delta,
compensation does not happen (because dst_incr == ideal_dst_incr)
but compensation_distance is set
regression since 01ebb57c03
Found-by: wm4 <nfxjfg@googlemail.com>
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
so tsf option in aresample will have effect
previously tsf/internal_sample_format had no effect
fate is updated
s32p previously used fltp internally
dblp previously used fltp/dblp internally
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
except filter_length == 1
odd filter_length gives worse frequency response,
even when compared with shorter filter_length
also makes build_filter simpler
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
integrate it inside multiple_resample
allow some calculations to be performed outside loop
Suggested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
This is faster 2871 -> 2189 cycles for int16 matrixbench -> 23456hz
Fixes a integer overflow in a artificial corner case
Fixes part of 668007-media
Found-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes undefined operation
Fixes part of 668007-media
Found-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
give very bad quality for soxr resampler.
linear_interp is intended for using linear interpolation
between filter bank so quality will be better.
i guess this is misunderstood as 'do not use filter bank,
but directly interpolate linearly between samples'.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
separate dsp.resample to dsp.resample_common and dsp.resample_linear
and choose to call faster resample_common even when linear_interp=on
when c->frac and c->dst_incr_mod are both zero
speed up resampling when exact_rational and linear_interp are both
enabled because exact_rational force c->frac and c->dst_incr_mod to
be zero when soft compensation does not happen
benchmark on exact_rational=on:linear_interp=on
old new
real 8.432s 5.097s
user 7.679s 4.989s
sys 0.125s 0.107s
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
high phase_count is only useful when dst_incr_mod is non zero
in other word, it is only useful on soft compensation
on init, it will build filter with low phase_count
but when soft compensation is enabled, rebuild filter
with high phase_count
this approach saves lots of memory
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
because exact_rational does not guarantee
that phase_count is even
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
give high quality resampling
as good as with linear_interp=on
as fast as without linear_interp=on
tested visually with ffplay
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000, showcqt=gamma=5"
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000:linear_interp=on, showcqt=gamma=5"
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000:exact_rational=on, showcqt=gamma=5"
slightly speed improvement
for fair comparison with -cpuflags 0
audio.wav is ~ 1 hour 44100 stereo 16bit wav file
ffmpeg -i audio.wav -af aresample=osr=48000 -f null -
old new
real 13.498s 13.121s
user 13.364s 12.987s
sys 0.131s 0.129s
linear_interp=on
old new
real 23.035s 23.050s
user 22.907s 22.917s
sys 0.119s 0.125s
exact_rational=on
real 12.418s
user 12.298s
sys 0.114s
possibility to decrease memory usage if soft compensation is ignored
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
This fixes the sum of the integer coefficients ending up summing to a value
larger than the value representing unity.
This issue occurs with qN0.dts when converting to stereo
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Michael Niedermayer
hexpointer
Diego Biurrun
James Almer
Muhammad Faiz
Carl Eugen Hoyos
Matthieu Bouron
Vodyannikov Aleksandr
Paul B Mahol
Timothy Gu
Clément Bœsch