Removed +len1 in call to s->mix_2_1_f() as I found no logical explanation for it. After removal, problem was gone.
Signed-off-by: Hendrik Schreiber <hs@tagtraum.com>
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>
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>
Fixes undefined operation
Fixes part of 668007-media
Found-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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>
or if no rematrix and no resampling is performed and the input is 16bit
note reampling and rematrix itself always use more than 16bit internally
the "internal" sampling format is the format between these steps
Its unlikely the difference from this commit is audible in any case
unless there is some bug either before or after the change.
but multiple people prefer this and it slightly improves the precission
of computations.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids leaks if the user doest call swr_close() after a failed init
Found-by: James Almer <jamrial@gmail.com>
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previous version reviewed-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Previous version reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix crash when doing 8 ch conversion from apps compiled with MSVS
Thanks to Ronald for giving this hint:
https://ffmpeg.org/pipermail/ffmpeg-devel/2015-May/173049.html
Reviewed-by: "Ronald S. Bultje" <rsbultje@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The constant may change in libavutil but the library may be compiled
against an older version, thus rejecting a value which is otherwise
supported by the new libavutil.
INT_MAX is used here to denote the max allowed value for a sample format.
The opt-test code is changed to provide a valid reference example.
