Also bump the minor versions of all libraries, to signify the
API change of splitting the version.h headers and adding the
new version_major.h header.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is done a second time for 5.0 because master was
merged into 5.0 so that it contains the recent DOVI additions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Instead only include libavutil/version.h; including avutil.h is a
remnant from the time in which the version was in it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
* commit '07a2b155949eb267cdfc7805f42c7b3375f9c7c5':
Bump major versions of all libraries
A few API deprecated ~2 years ago or more are also postponed here for
varying reasons.
FF_API_LOWRES:
Since this functionality depends on AVStream->codec, i figure the two can
be removed at the same time in the next bump or so.
FF_API_AVCTX_TIMEBASE:
Couldn't get this one to work. Not just libavcodec but apparently also
libavformat and ffmpeg.c expect AVCodecContext->time_base to be set for
decoding. Upon removal some tests report a different generic stream time
base (like 1/25), and others lose packet duration values. I guess it's
somehow tied to the AVStream->codec clusterfuck.
It can be dealt with alongside FF_API_LAVF_AVCTX in the next bump.
FF_API_OLD_FILTER_OPTS_ERROR:
This one is meant to remain after FF_API_OLD_FILTER_OPTS is removed.
Its purpose is displaying the corrected command line using the new syntax
as a suggestion as part of the error message.
Merged-by: James Almer <jamrial@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>
Kaiser windows inherently don't require beta to be an integer. This was
an arbitrary restriction. Moreover, soxr does not require it, and in
fact often estimates beta to a non-integral value.
Thus, this patch allows greater flexibility for swresample clients.
Micro version is updated.
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
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>
Based on commit fb1ddcdc8f by Luca Barbato <lu_zero@gentoo.org>
Adapted for libswresample by Michael Niedermayer
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
