| Message ID | 20200611161227.5622-7-onemda@gmail.com |
|---|---|
| State | New |
| Headers | show |
| Series | [FFmpeg-devel,1/7] avfilter/af_biquads: switch to activate() | expand |
| Context | Check | Description |
|---|---|---|
| andriy/default | pending | |
| andriy/make | success | Make finished |
| andriy/make_fate | success | Make fate finished |
Paul B Mahol (12020-06-11): > Signed-off-by: Paul B Mahol <onemda@gmail.com> > --- > doc/filters.texi | 60 ++ > libavfilter/Makefile | 1 + > libavfilter/af_afwtdn.c | 1345 ++++++++++++++++++++++++++++++++++++++ > libavfilter/allfilters.c | 1 + > 4 files changed, 1407 insertions(+) > create mode 100644 libavfilter/af_afwtdn.c I still oppose to this filter on the basis of the name. > > diff --git a/doc/filters.texi b/doc/filters.texi > index c2960e33c7..d89ebc5122 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -1314,6 +1314,66 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo > aformat=sample_fmts=u8|s16:channel_layouts=stereo > @end example > > +@section afwtdn > +Reduce broadband noise from input samples using Wavelets. Please document in user-oriented terms and not in developer-oriented the benefits of this filter for the user. You need to explain WHAT it does different from the n other denoisers, not HOW it does it. > + > +A description of the accepted options follows. > + > +@table @option > +@item sigma > +Set the noise sigma, allowed range is from 0 to 1. > +Default value is 0. > +This option controls strength of denoising applied to input samples. > +Most useful way to set this option is via decibels, eg. -45dB. > + > +@item levels > +Set the number of wavelet levels of decomposition. > +Allowed range is from 1 to 12. > +Default value is 10. > +Setting this too low make denoising performance very poor. > + > +@item wavet > +Set wavelet type for decomposition of input frame. > +They are sorted by number of coefficients, from lowest to highest. > +More coefficients means worse filtering speed, but overall better quality. > +Available wavelets are: > + > +@table @samp > +@item sym2 > +@item sym4 > +@item rbior68 > +@item deb10 > +@item sym10 > +@item coif5 > +@item bl3 > +@end table > + > +@item percent > +Set percent of full denoising. Allowed range is from 0 to 100 percent. > +Default value is 85 percent or partial denoising. > + > +@item profile > +If enabled, first input frame will be used as noise profile. > +If first frame samples contain non-noise performance will be very poor. > + > +@item adaptive > +If enabled, input frames are analyzed for presence of noise. > +If noise is detected with high possibility then input frame profile will be > +used for processing following frames, until new noise frame is detected. > + > +@item samples > +Set size of single frame in number of samples. Allowed range is from 512 to > +65536. Default frame size is 8192 samples. > + > +@item softness > +Set softness applied inside thresholding function. Allowed range is from 0 to > +10. Default softness is 1. > +@end table > + > +@subsection Commands > + > +This filter supports subset of its options as @ref{commands}. > + > @section agate > > A gate is mainly used to reduce lower parts of a signal. This kind of signal > diff --git a/libavfilter/Makefile b/libavfilter/Makefile > index 5123540653..191826a622 100644 > --- a/libavfilter/Makefile > +++ b/libavfilter/Makefile > @@ -50,6 +50,7 @@ OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o > OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o > OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o > OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o > +OBJS-$(CONFIG_AFWTDN_FILTER) += af_afwtdn.o > OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o > OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o > OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o > diff --git a/libavfilter/af_afwtdn.c b/libavfilter/af_afwtdn.c > new file mode 100644 > index 0000000000..d2793d4d92 > --- /dev/null > +++ b/libavfilter/af_afwtdn.c > @@ -0,0 +1,1345 @@ > +/* > + * Copyright (c) 2020 Paul B Mahol > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +#include <float.h> > + > +#include "libavutil/avassert.h" > +#include "libavutil/avstring.h" > +#include "libavutil/opt.h" > +#include "avfilter.h" > +#include "audio.h" > +#include "filters.h" > +#include "formats.h" > + > +enum WaveletTypes { > + SYM2, > + SYM4, > + RBIOR68, > + DEB10, > + SYM10, > + COIF5, > + BL3, > + NB_WAVELET_TYPES, > +}; > + > +static const double bl3_lp[42] = { > + 0.000146098, -0.000232304, -0.000285414, 0.000462093, 0.000559952, > + -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426, > + -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331, > + 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147, > + 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869, > + 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477, > + -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748, > + -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304, > + 0.000146098, 0.0, > +}; > + > +static const double bl3_hp[42] = { > + 0.0, 0.000146098, 0.000232304, -0.000285414, -0.000462093, 0.000559952, > + 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426, > + -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331, > + -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147, > + 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869, > + -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477, > + -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748, > + 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304, > + 0.000146098, > +}; > + > +static const double bl3_ilp[42] = { > + 0.0, 0.000146098, -0.000232304, -0.000285414, 0.000462093, 0.000559952, > + -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426, > + -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331, > + 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147, > + 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869, > + 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477, > + -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748, > + -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304, > + 0.000146098, > +}; > + > +static const double bl3_ihp[42] = { > + 0.000146098, 0.000232304, -0.000285414, -0.000462093, 0.000559952, > + 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426, > + -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331, > + -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147, > + 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869, > + -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477, > + -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748, > + 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304, > + 0.000146098, > +}; > + > +static const double sym10_lp[20] = { > + 0.0007701598091144901, 9.563267072289475e-05, > + -0.008641299277022422, -0.0014653825813050513, > + 0.0459272392310922, 0.011609893903711381, > + -0.15949427888491757, -0.07088053578324385, > + 0.47169066693843925, 0.7695100370211071, > + 0.38382676106708546, -0.03553674047381755, > + -0.0319900568824278, 0.04999497207737669, > + 0.005764912033581909, -0.02035493981231129, > + -0.0008043589320165449, 0.004593173585311828, > + 5.7036083618494284e-05, -0.0004593294210046588, > +}; > + > +static const double sym10_hp[20] = { > + 0.0004593294210046588, 5.7036083618494284e-05, > + -0.004593173585311828, -0.0008043589320165449, > + 0.02035493981231129, 0.005764912033581909, > + -0.04999497207737669, -0.0319900568824278, > + 0.03553674047381755, 0.38382676106708546, > + -0.7695100370211071, 0.47169066693843925, > + 0.07088053578324385, -0.15949427888491757, > + -0.011609893903711381, 0.0459272392310922, > + 0.0014653825813050513, -0.008641299277022422, > + -9.563267072289475e-05, 0.0007701598091144901, > +}; > + > +static const double sym10_ilp[20] = { > + -0.0004593294210046588, 5.7036083618494284e-05, > + 0.004593173585311828, -0.0008043589320165449, > + -0.02035493981231129, 0.005764912033581909, > + 0.04999497207737669, -0.0319900568824278, > + -0.03553674047381755, 0.38382676106708546, > + 0.7695100370211071, 0.47169066693843925, > + -0.07088053578324385, -0.15949427888491757, > + 0.011609893903711381, 0.0459272392310922, > + -0.0014653825813050513, -0.008641299277022422, > + 9.563267072289475e-05, 0.0007701598091144901, > +}; > + > +static const double sym10_ihp[20] = { > + 0.0007701598091144901, -9.563267072289475e-05, > + -0.008641299277022422, 0.0014653825813050513, > + 0.0459272392310922, -0.011609893903711381, > + -0.15949427888491757, 0.07088053578324385, > + 0.47169066693843925, -0.7695100370211071, > + 0.38382676106708546, 0.03553674047381755, > + -0.0319900568824278, -0.04999497207737669, > + 0.005764912033581909, 0.02035493981231129, > + -0.0008043589320165449, -0.004593173585311828, > + 5.7036083618494284e-05, 0.0004593294210046588, > +}; > + > +static const double rbior68_lp[18] = { > + 0.0, 0.0, 0.0, 0.0, > + 0.014426282505624435, 0.014467504896790148, > + -0.07872200106262882, -0.04036797903033992, > + 0.41784910915027457, 0.7589077294536541, > + 0.41784910915027457, -0.04036797903033992, > + -0.07872200106262882, 0.014467504896790148, > + 0.014426282505624435, 0.0, 0.0, 0.0, > +}; > + > +static const double rbior68_hp[18] = { > + -0.0019088317364812906, -0.0019142861290887667, > + 0.016990639867602342, 0.01193456527972926, > + -0.04973290349094079, -0.07726317316720414, > + 0.09405920349573646, 0.4207962846098268, > + -0.8259229974584023, 0.4207962846098268, > + 0.09405920349573646, -0.07726317316720414, > + -0.04973290349094079, 0.01193456527972926, > + 0.016990639867602342, -0.0019142861290887667, > + -0.0019088317364812906, 0.0, > +}; > + > +static const double rbior68_ilp[18] = { > + 0.0019088317364812906, -0.0019142861290887667, > + -0.016990639867602342, 0.01193456527972926, > + 0.04973290349094079, -0.07726317316720414, > + -0.09405920349573646, 0.4207962846098268, > + 0.8259229974584023, 0.4207962846098268, > + -0.09405920349573646, -0.07726317316720414, > + 0.04973290349094079, 0.01193456527972926, > + -0.016990639867602342, -0.0019142861290887667, > + 0.0019088317364812906, 0.0, > +}; > + > +static const double rbior68_ihp[18] = { > + 0.0, 0.0, 0.0, 0.0, > + 0.014426282505624435, -0.014467504896790148, > + -0.07872200106262882, 0.04036797903033992, > + 0.41784910915027457, -0.7589077294536541, > + 0.41784910915027457, 0.04036797903033992, > + -0.07872200106262882, -0.014467504896790148, > + 0.014426282505624435, 0.0, 0.0, 0.0, > +}; > + > +static const double coif5_lp[30] = { > + -9.517657273819165e-08, -1.6744288576823017e-07, > + 2.0637618513646814e-06, 3.7346551751414047e-06, > + -2.1315026809955787e-05, -4.134043227251251e-05, > + 0.00014054114970203437, 0.00030225958181306315, > + -0.0006381313430451114, -0.0016628637020130838, > + 0.0024333732126576722, 0.006764185448053083, > + -0.009164231162481846, -0.01976177894257264, > + 0.03268357426711183, 0.0412892087501817, > + -0.10557420870333893, -0.06203596396290357, > + 0.4379916261718371, 0.7742896036529562, > + 0.4215662066908515, -0.05204316317624377, > + -0.09192001055969624, 0.02816802897093635, > + 0.023408156785839195, -0.010131117519849788, > + -0.004159358781386048, 0.0021782363581090178, > + 0.00035858968789573785, -0.00021208083980379827, > +}; > + > +static const double coif5_hp[30] = { > + 0.00021208083980379827, 0.00035858968789573785, > + -0.0021782363581090178, -0.004159358781386048, > + 0.010131117519849788, 0.023408156785839195, > + -0.02816802897093635, -0.09192001055969624, > + 0.05204316317624377, 0.4215662066908515, > + -0.7742896036529562, 0.4379916261718371, > + 0.06203596396290357, -0.10557420870333893, > + -0.0412892087501817, 0.03268357426711183, > + 0.01976177894257264, -0.009164231162481846, > + -0.006764185448053083, 0.0024333732126576722, > + 0.0016628637020130838, -0.0006381313430451114, > + -0.00030225958181306315, 0.00014054114970203437, > + 4.134043227251251e-05, -2.1315026809955787e-05, > + -3.7346551751414047e-06, 2.0637618513646814e-06, > + 1.6744288576823017e-07, -9.517657273819165e-08, > +}; > + > +static const double coif5_ilp[30] = { > + -0.00021208083980379827, 0.00035858968789573785, > + 0.0021782363581090178, -0.004159358781386048, > + -0.010131117519849788, 0.023408156785839195, > + 0.02816802897093635, -0.09192001055969624, > + -0.05204316317624377, 0.4215662066908515, > + 0.7742896036529562, 0.4379916261718371, > + -0.06203596396290357, -0.10557420870333893, > + 0.0412892087501817, 0.03268357426711183, > + -0.01976177894257264, -0.009164231162481846, > + 0.006764185448053083, 0.0024333732126576722, > + -0.0016628637020130838, -0.0006381313430451114, > + 0.00030225958181306315, 0.00014054114970203437, > + -4.134043227251251e-05, -2.1315026809955787e-05, > + 3.7346551751414047e-06, 2.0637618513646814e-06, > + -1.6744288576823017e-07, -9.517657273819165e-08, > +}; > + > +static const double coif5_ihp[30] = { > + -9.517657273819165e-08, 1.6744288576823017e-07, > + 2.0637618513646814e-06, -3.7346551751414047e-06, > + -2.1315026809955787e-05, 4.134043227251251e-05, > + 0.00014054114970203437, -0.00030225958181306315, > + -0.0006381313430451114, 0.0016628637020130838, > + 0.0024333732126576722, -0.006764185448053083, > + -0.009164231162481846, 0.01976177894257264, > + 0.03268357426711183, -0.0412892087501817, > + -0.10557420870333893, 0.06203596396290357, > + 0.4379916261718371, -0.7742896036529562, > + 0.4215662066908515, 0.05204316317624377, > + -0.09192001055969624, -0.02816802897093635, > + 0.023408156785839195, 0.010131117519849788, > + -0.004159358781386048, -0.0021782363581090178, > + 0.00035858968789573785, 0.00021208083980379827, > +}; > + > +static const double deb10_lp[20] = { > + -1.326420300235487e-05, 9.358867000108985e-05, > + -0.0001164668549943862, -0.0006858566950046825, > + 0.00199240529499085, 0.0013953517469940798, > + -0.010733175482979604, 0.0036065535669883944, > + 0.03321267405893324, -0.02945753682194567, > + -0.07139414716586077, 0.09305736460380659, > + 0.12736934033574265, -0.19594627437659665, > + -0.24984642432648865, 0.2811723436604265, > + 0.6884590394525921, 0.5272011889309198, > + 0.18817680007762133, 0.026670057900950818, > +}; > + > +static const double deb10_hp[20] = { > + -0.026670057900950818, 0.18817680007762133, > + -0.5272011889309198, 0.6884590394525921, > + -0.2811723436604265, -0.24984642432648865, > + 0.19594627437659665, 0.12736934033574265, > + -0.09305736460380659, -0.07139414716586077, > + 0.02945753682194567, 0.03321267405893324, > + -0.0036065535669883944, -0.010733175482979604, > + -0.0013953517469940798, 0.00199240529499085, > + 0.0006858566950046825, -0.0001164668549943862, > + -9.358867000108985e-05, -1.326420300235487e-05, > +}; > + > +static const double deb10_ilp[20] = { > + 0.026670057900950818, 0.18817680007762133, > + 0.5272011889309198, 0.6884590394525921, > + 0.2811723436604265, -0.24984642432648865, > + -0.19594627437659665, 0.12736934033574265, > + 0.09305736460380659, -0.07139414716586077, > + -0.02945753682194567, 0.03321267405893324, > + 0.0036065535669883944, -0.010733175482979604, > + 0.0013953517469940798, 0.00199240529499085, > + -0.0006858566950046825, -0.0001164668549943862, > + 9.358867000108985e-05, -1.326420300235487e-05, > +}; > + > +static const double deb10_ihp[20] = { > + -1.326420300235487e-05, -9.358867000108985e-05, > + -0.0001164668549943862, 0.0006858566950046825, > + 0.00199240529499085, -0.0013953517469940798, > + -0.010733175482979604, -0.0036065535669883944, > + 0.03321267405893324, 0.02945753682194567, > + -0.07139414716586077, -0.09305736460380659, > + 0.12736934033574265, 0.19594627437659665, > + -0.24984642432648865, -0.2811723436604265, > + 0.6884590394525921, -0.5272011889309198, > + 0.18817680007762133, -0.026670057900950818, > +}; > + > +static const double sym4_lp[8] = { > + -0.07576571478927333, > + -0.02963552764599851, > + 0.49761866763201545, > + 0.8037387518059161, > + 0.29785779560527736, > + -0.09921954357684722, > + -0.012603967262037833, > + 0.0322231006040427, > +}; > + > +static const double sym4_hp[8] = { > + -0.0322231006040427, > + -0.012603967262037833, > + 0.09921954357684722, > + 0.29785779560527736, > + -0.8037387518059161, > + 0.49761866763201545, > + 0.02963552764599851, > + -0.07576571478927333, > +}; > + > +static const double sym4_ilp[8] = { > + 0.0322231006040427, > + -0.012603967262037833, > + -0.09921954357684722, > + 0.29785779560527736, > + 0.8037387518059161, > + 0.49761866763201545, > + -0.02963552764599851, > + -0.07576571478927333, > +}; > + > +static const double sym4_ihp[8] = { > + -0.07576571478927333, > + 0.02963552764599851, > + 0.49761866763201545, > + -0.8037387518059161, > + 0.29785779560527736, > + 0.09921954357684722, > + -0.012603967262037833, > + -0.0322231006040427, > +}; > + > +static const double sym2_lp[4] = { > + -0.12940952255092145, 0.22414386804185735, > + 0.836516303737469, 0.48296291314469025, > +}; > + > +static const double sym2_hp[4] = { > + -0.48296291314469025, 0.836516303737469, > + -0.22414386804185735, -0.12940952255092145, > +}; > + > +static const double sym2_ilp[4] = { > + 0.48296291314469025, 0.836516303737469, > + 0.22414386804185735, -0.12940952255092145, > +}; > + > +static const double sym2_ihp[4] = { > + -0.12940952255092145, -0.22414386804185735, > + 0.836516303737469, -0.48296291314469025, > +}; You did not compute these numbers in your head nor did you type them by hand: they are not source code. We must include the whole source code. > + > +#define MAX_LEVELS 13 > + > +typedef struct ChannelParams { > + int *output_length; > + int *filter_length; > + double **output_coefs; > + double **subbands_to_free; > + double **filter_coefs; > + > + int tempa_length; > + int tempa_len_max; > + int temp_in_length; > + int temp_in_max_length; > + int buffer_length; > + int min_left_ext; > + int max_left_ext; > + > + double *tempa; > + double *tempd; > + double *temp_in; > + double *buffer; > + double *buffer2; > + double *prev; > + double *overlap; > +} ChannelParams; > + > +typedef struct AudioFWTDNContext { > + const AVClass *class; > + > + double sigma; > + double percent; > + double softness; > + > + uint64_t sn; > + int64_t eof_pts; > + > + int wavelet_type; > + int channels; > + int nb_samples; > + int levels; > + int wavelet_length; > + int need_profile; > + int got_profile; > + int adaptive; > + > + int delay; > + int drop_samples; > + int padd_samples; > + int overlap_length; > + int prev_length; > + ChannelParams *cp; > + > + const double *lp, *hp; > + const double *ilp, *ihp; > + > + AVFrame *stddev, *absmean, *filter; > + AVFrame *new_stddev, *new_absmean; > + > + int (*filter_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs); > +} AudioFWTDNContext; > + > +#define OFFSET(x) offsetof(AudioFWTDNContext, x) > +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM > +#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM > + > +static const AVOption afwtdn_options[] = { > + { "sigma", "set noise sigma", OFFSET(sigma), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, AFR }, > + { "levels", "set number of wavelet levels", OFFSET(levels), AV_OPT_TYPE_INT, {.i64=10}, 1, MAX_LEVELS-1, AF }, > + { "wavet", "set wavelet type", OFFSET(wavelet_type), AV_OPT_TYPE_INT, {.i64=SYM10}, 0, NB_WAVELET_TYPES - 1, AF, "wavet" }, > + { "sym2", "sym2", 0, AV_OPT_TYPE_CONST, {.i64=SYM2}, 0, 0, AF, "wavet" }, > + { "sym4", "sym4", 0, AV_OPT_TYPE_CONST, {.i64=SYM4}, 0, 0, AF, "wavet" }, > + { "rbior68", "rbior68", 0, AV_OPT_TYPE_CONST, {.i64=RBIOR68}, 0, 0, AF, "wavet" }, > + { "deb10", "deb10", 0, AV_OPT_TYPE_CONST, {.i64=DEB10}, 0, 0, AF, "wavet" }, > + { "sym10", "sym10", 0, AV_OPT_TYPE_CONST, {.i64=SYM10}, 0, 0, AF, "wavet" }, > + { "coif5", "coif5", 0, AV_OPT_TYPE_CONST, {.i64=COIF5}, 0, 0, AF, "wavet" }, > + { "bl3", "bl3", 0, AV_OPT_TYPE_CONST, {.i64=BL3}, 0, 0, AF, "wavet" }, > + { "percent", "set percent of full denoising", OFFSET(percent),AV_OPT_TYPE_DOUBLE, {.dbl=85}, 0, 100, AFR }, > + { "profile", "profile noise", OFFSET(need_profile), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AFR }, > + { "adaptive", "adaptive profiling of noise", OFFSET(adaptive), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AFR }, > + { "samples", "set frame size in number of samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=8192}, 512, 65536, AF }, > + { "softness", "set thresholding softness", OFFSET(softness), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 10, AFR }, > + { NULL } > +}; > + > +AVFILTER_DEFINE_CLASS(afwtdn); > + > +static int query_formats(AVFilterContext *ctx) > +{ > + AVFilterFormats *formats = NULL; > + AVFilterChannelLayouts *layouts = NULL; > + static const enum AVSampleFormat sample_fmts[] = { > + AV_SAMPLE_FMT_DBLP, > + AV_SAMPLE_FMT_NONE > + }; > + int ret; > + > + formats = ff_make_format_list(sample_fmts); > + if (!formats) > + return AVERROR(ENOMEM); > + ret = ff_set_common_formats(ctx, formats); > + if (ret < 0) > + return ret; > + > + layouts = ff_all_channel_counts(); > + if (!layouts) > + return AVERROR(ENOMEM); > + > + ret = ff_set_common_channel_layouts(ctx, layouts); > + if (ret < 0) > + return ret; > + > + formats = ff_all_samplerates(); > + return ff_set_common_samplerates(ctx, formats); > +} > + > +#define pow2(x) (1U << (x)) > +#define mod_pow2(x, power_of_two) ((x) & ((power_of_two) - 1)) > + > +static void conv_down(double *in, int in_length, double *low, double *high, > + int out_length, const double *lp, const double *hp, > + int wavelet_length, int skip, > + double *buffer, int buffer_length) > +{ > + double thigh = 0.0, tlow = 0.0; > + int buff_idx = 1 + skip; > + > + memcpy(buffer, in, buff_idx * sizeof(*buffer)); > + memset(buffer + buff_idx, 0, (buffer_length - buff_idx) * sizeof(*buffer)); > + > + for (int i = 0; i < out_length - 1; i++) { > + double thigh = 0.0, tlow = 0.0; > + > + for (int j = 0; j < wavelet_length; j++) { > + const int idx = mod_pow2(-j + buff_idx - 1, buffer_length); > + const double btemp = buffer[idx]; > + > + thigh += btemp * hp[j]; > + tlow += btemp * lp[j]; > + } > + > + high[i] = thigh; > + low[i] = tlow; > + buffer[buff_idx++] = in[2 * i + 1 + skip]; > + buffer[buff_idx++] = in[2 * i + 2 + skip]; > + buff_idx = mod_pow2(buff_idx, buffer_length); > + } > + > + for (int i = 0; i < wavelet_length; i++) { > + const int idx = mod_pow2(-i + buff_idx - 1, buffer_length); > + const double btemp = buffer[idx]; > + > + thigh += btemp * hp[i]; > + tlow += btemp * lp[i]; > + } > + > + high[out_length - 1] = thigh; > + low[out_length - 1] = tlow; > +} > + > +static int left_ext(int wavelet_length, int levels, uint64_t sn) > +{ > + if (!sn) > + return 0; > + return (pow2(levels) - 1) * (wavelet_length - 2) + mod_pow2(sn, pow2(levels)); > +} > + > +static int nb_coefs(int length, int level, uint64_t sn) > +{ > + const int pow2_level = pow2(level); > + > + return (sn + length) / pow2_level - sn / pow2_level; > +} > + > +static int reallocate_inputs(double **out, int *out_length, > + int in_length, int levels, int ch, uint64_t sn) > +{ > + const int temp_length = nb_coefs(in_length, levels, sn); > + > + for (int level = 0; level < levels; level++) { > + const int temp_length = nb_coefs(in_length, level + 1, sn); > + > + if (temp_length > out_length[level]) { > + av_freep(&out[level]); > + out_length[level] = 0; > + > + out[level] = av_calloc(temp_length + 1, sizeof(**out)); > + if (!out[level]) > + return AVERROR(ENOMEM); > + out_length[level] = temp_length + 1; > + } > + > + memset(out[level] + temp_length, 0, > + (out_length[level] - temp_length) * sizeof(**out)); > + out_length[level] = temp_length; > + } > + > + if (temp_length > out_length[levels]) { > + av_freep(&out[levels]); > + out_length[levels] = 0; > + > + out[levels] = av_calloc(temp_length + 1, sizeof(**out)); > + if (!out[levels]) > + return AVERROR(ENOMEM); > + out_length[levels] = temp_length + 1; > + } > + > + memset(out[levels] + temp_length, 0, > + (out_length[levels] - temp_length) * sizeof(**out)); > + out_length[levels] = temp_length; > + > + return 0; > +} > + > +static int max_left_zeros_inverse(int levels, int level, int wavelet_length) > +{ > + return (pow2(levels - level) - 1) * (wavelet_length - 1); > +} > + > +static int reallocate_outputs(AudioFWTDNContext *s, > + double **out, int *out_length, > + int in_length, int levels, int ch, uint64_t sn) > +{ > + ChannelParams *cp = &s->cp[ch]; > + int temp_length = 0; > + int add = 0; > + > + for (int level = 0; level < levels; level++) { > + temp_length = nb_coefs(in_length, level + 1, sn); > + if (temp_length > out_length[level]) { > + av_freep(&cp->subbands_to_free[level]); > + out_length[level] = 0; > + > + add = max_left_zeros_inverse(levels, level + 1, s->wavelet_length); > + cp->subbands_to_free[level] = av_calloc(add + temp_length + 1, sizeof(**out)); > + if (!cp->subbands_to_free[level]) > + return AVERROR(ENOMEM); > + out_length[level] = add + temp_length + 1; > + out[level] = cp->subbands_to_free[level] + add; > + } > + > + memset(out[level] + temp_length, 0, > + FFMAX(out_length[level] - temp_length - add, 0) * sizeof(**out)); > + out_length[level] = temp_length; > + } > + > + temp_length = nb_coefs(in_length, levels, sn); > + if (temp_length > out_length[levels]) { > + av_freep(&cp->subbands_to_free[levels]); > + out_length[levels] = 0; > + > + cp->subbands_to_free[levels] = av_calloc(temp_length + 1, sizeof(**out)); > + if (!cp->subbands_to_free[levels]) > + return AVERROR(ENOMEM); > + out_length[levels] = temp_length + 1; > + out[levels] = cp->subbands_to_free[levels]; > + } > + > + memset(out[levels] + temp_length, 0, > + (out_length[levels] - temp_length) * sizeof(**out)); > + out_length[levels] = temp_length; > + > + return 0; > +} > + > +static int discard_left_ext(int wavelet_length, int levels, int level, uint64_t sn) > +{ > + if (levels == level || sn == 0) > + return 0; > + return (pow2(levels - level) - 1) * (wavelet_length - 2) + mod_pow2(sn, pow2(levels)) / pow2(level); > +} > + > +static int forward(AudioFWTDNContext *s, > + const double *in, int in_length, > + double **out, int *out_length, int ch, uint64_t sn) > +{ > + ChannelParams *cp = &s->cp[ch]; > + int levels = s->levels; > + int skip = sn ? s->wavelet_length - 1 : 1; > + int leftext, ret; > + > + ret = reallocate_inputs(out, out_length, in_length, levels, ch, sn); > + if (ret < 0) > + return ret; > + ret = reallocate_outputs(s, cp->filter_coefs, cp->filter_length, > + in_length, levels, ch, sn); > + if (ret < 0) > + return ret; > + > + leftext = left_ext(s->wavelet_length, levels, sn); > + > + if (cp->temp_in_max_length < in_length + cp->max_left_ext + skip) { > + av_freep(&cp->temp_in); > + cp->temp_in_max_length = in_length + cp->max_left_ext + skip; > + cp->temp_in = av_calloc(cp->temp_in_max_length, sizeof(*cp->temp_in)); > + if (!cp->temp_in) { > + cp->temp_in_max_length = 0; > + return AVERROR(ENOMEM); > + } > + } > + > + memset(cp->temp_in, 0, cp->temp_in_max_length * sizeof(*cp->temp_in)); > + cp->temp_in_length = in_length + leftext; > + > + if (leftext) > + memcpy(cp->temp_in, cp->prev + s->prev_length - leftext, leftext * sizeof(*cp->temp_in)); > + memcpy(cp->temp_in + leftext, in, in_length * sizeof(*in)); > + > + if (levels == 1) { > + conv_down(cp->temp_in, cp->temp_in_length, out[1], out[0], out_length[1], > + s->lp, s->hp, s->wavelet_length, skip, > + cp->buffer, cp->buffer_length); > + } else { > + int discard = discard_left_ext(s->wavelet_length, levels, 1, sn); > + int tempa_length_prev; > + > + if (cp->tempa_len_max < (in_length + cp->max_left_ext + s->wavelet_length - 1) / 2) { > + av_freep(&cp->tempa); > + av_freep(&cp->tempd); > + cp->tempa_len_max = (in_length + cp->max_left_ext + s->wavelet_length - 1) / 2; > + cp->tempa = av_calloc(cp->tempa_len_max, sizeof(*cp->tempa)); > + cp->tempd = av_calloc(cp->tempa_len_max, sizeof(*cp->tempd)); > + if (!cp->tempa || !cp->tempd) { > + cp->tempa_len_max = 0; > + return AVERROR(ENOMEM); > + } > + } > + > + memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa)); > + memset(cp->tempd, 0, cp->tempa_len_max * sizeof(*cp->tempd)); > + > + cp->tempa_length = out_length[0] + discard; > + conv_down(cp->temp_in, cp->temp_in_length, > + cp->tempa, cp->tempd, cp->tempa_length, > + s->lp, s->hp, s->wavelet_length, skip, > + cp->buffer, cp->buffer_length); > + memcpy(out[0], cp->tempd + discard, out_length[0] * sizeof(**out)); > + tempa_length_prev = cp->tempa_length; > + > + for (int level = 1; level < levels - 1; level++) { > + if (out_length[level] == 0) > + return 0; > + discard = discard_left_ext(s->wavelet_length, levels, level + 1, sn); > + cp->tempa_length = out_length[level] + discard; > + conv_down(cp->tempa, tempa_length_prev, > + cp->tempa, cp->tempd, cp->tempa_length, > + s->lp, s->hp, s->wavelet_length, skip, > + cp->buffer, cp->buffer_length); > + memcpy(out[level], cp->tempd + discard, out_length[level] * sizeof(**out)); > + tempa_length_prev = cp->tempa_length; > + } > + > + if (out_length[levels] == 0) > + return 0; > + conv_down(cp->tempa, cp->tempa_length, out[levels], out[levels - 1], out_length[levels], > + s->lp, s->hp, s->wavelet_length, skip, > + cp->buffer, cp->buffer_length); > + } > + > + if (s->prev_length < in_length) { > + memcpy(cp->prev, in + in_length - cp->max_left_ext, cp->max_left_ext * sizeof(*cp->prev)); > + } else { > + memmove(cp->prev, cp->prev + in_length, (s->prev_length - in_length) * sizeof(*cp->prev)); > + memcpy(cp->prev + s->prev_length - in_length, in, in_length * sizeof(*cp->prev)); > + } > + > + return 0; > +} > + > +static void conv_up(double *low, double *high, int in_length, double *out, int out_length, > + const double *lp, const double *hp, int filter_length, > + double *buffer, double *buffer2, int buffer_length) > +{ > + int shift = 0, buff_idx = 0, in_idx = 0; > + > + memset(buffer, 0, buffer_length * sizeof(*buffer)); > + memset(buffer2, 0, buffer_length * sizeof(*buffer2)); > + > + for (int i = 0; i < out_length; i++) { > + double sum = 0.0; > + > + if ((i & 1) == 0) { > + if (in_idx < in_length) { > + buffer[buff_idx] = low[in_idx]; > + buffer2[buff_idx] = high[in_idx++]; > + } else { > + buffer[buff_idx] = 0; > + buffer2[buff_idx] = 0; > + } > + buff_idx++; > + if (buff_idx >= buffer_length) > + buff_idx = 0; > + shift = 0; > + } > + > + for (int j = 0; j < (filter_length - shift + 1) / 2; j++) { > + const int idx = mod_pow2(-j + buff_idx - 1, buffer_length); > + > + sum += buffer[idx] * lp[j * 2 + shift] + buffer2[idx] * hp[j * 2 + shift]; > + } > + out[i] = sum; > + shift = 1; > + } > +} > + > +static int append_left_ext(int wavelet_length, int levels, int level, uint64_t sn) > +{ > + if (levels == level) > + return 0; > + > + return (pow2(levels - level) - 1) * (wavelet_length - 2) + > + mod_pow2(sn, pow2(levels)) / pow2(level); > +} > + > +static int inverse(AudioFWTDNContext *s, > + double **in, int *in_length, > + double *out, int out_length, int ch, uint64_t sn) > +{ > + ChannelParams *cp = &s->cp[ch]; > + const int levels = s->levels; > + int leftext = left_ext(s->wavelet_length, levels, sn); > + int temp_skip = 0; > + > + if (sn == 0) > + temp_skip = cp->min_left_ext; > + > + memset(out, 0, out_length * sizeof(*out)); > + > + if (cp->temp_in_max_length < out_length + cp->max_left_ext + s->wavelet_length - 1) { > + av_freep(&cp->temp_in); > + cp->temp_in_max_length = out_length + cp->max_left_ext + s->wavelet_length - 1; > + cp->temp_in = av_calloc(cp->temp_in_max_length, sizeof(*cp->temp_in)); > + if (!cp->temp_in) { > + cp->temp_in_max_length = 0; > + return AVERROR(ENOMEM); > + } > + } > + > + memset(cp->temp_in, 0, cp->temp_in_max_length * sizeof(*cp->temp_in)); > + cp->temp_in_length = out_length + cp->max_left_ext; > + > + if (levels == 1) { > + conv_up(in[1], in[0], in_length[1], cp->temp_in, cp->temp_in_length, > + s->ilp, s->ihp, s->wavelet_length, > + cp->buffer, cp->buffer2, cp->buffer_length); > + memcpy(out + cp->max_left_ext - leftext, cp->temp_in + temp_skip, > + FFMAX(0, out_length - (cp->max_left_ext - leftext)) * sizeof(*out)); > + } else { > + double *hp1, *hp2; > + int add, add2; > + > + if (cp->tempa_len_max < (out_length + cp->max_left_ext + s->wavelet_length - 1) / 2) { > + av_freep(&cp->tempa); > + cp->tempa_len_max = (out_length + cp->max_left_ext + s->wavelet_length - 1) / 2; > + cp->tempa = av_calloc(cp->tempa_len_max, sizeof(*cp->tempa)); > + if (!cp->tempa) { > + cp->tempa_len_max = 0; > + return AVERROR(ENOMEM); > + } > + } > + > + memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa)); > + > + hp1 = levels & 1 ? cp->temp_in : cp->tempa; > + hp2 = levels & 1 ? cp->tempa : cp->temp_in; > + > + add = append_left_ext(s->wavelet_length, levels, levels - 1, sn); > + conv_up(in[levels], in[levels - 1], in_length[levels], hp1, in_length[levels - 2] + add, > + s->ilp, s->ihp, s->wavelet_length, cp->buffer, cp->buffer2, cp->buffer_length); > + > + for (int level = levels - 1; level > 1; level--) { > + add2 = append_left_ext(s->wavelet_length, levels, level - 1, sn); > + add = append_left_ext(s->wavelet_length, levels, level, sn); > + conv_up(hp1, in[level - 1] - add, in_length[level - 1] + add, > + hp2, in_length[level - 2] + add2, > + s->ilp, s->ihp, s->wavelet_length, > + cp->buffer, cp->buffer2, cp->buffer_length); > + FFSWAP(double *, hp1, hp2); > + } > + > + add = append_left_ext(s->wavelet_length, levels, 1, sn); > + conv_up(hp1, in[0] - add, in_length[0] + add, cp->temp_in, cp->temp_in_length, > + s->ilp, s->ihp, s->wavelet_length, > + cp->buffer, cp->buffer2, cp->buffer_length); > + } > + > + memset(cp->temp_in, 0, temp_skip * sizeof(*cp->temp_in)); > + if (s->overlap_length <= out_length) { > + memcpy(out + cp->max_left_ext - leftext, cp->temp_in + temp_skip, > + FFMAX(0, out_length - (cp->max_left_ext - leftext)) * sizeof(*out)); > + for (int i = 0;i < FFMIN(s->overlap_length, out_length); i++) > + out[i] += cp->overlap[i]; > + > + memcpy(cp->overlap, cp->temp_in + out_length - (cp->max_left_ext - leftext), > + s->overlap_length * sizeof(*cp->overlap)); > + } else { > + for (int i = 0;i < s->overlap_length - (cp->max_left_ext - leftext); i++) > + cp->overlap[i + cp->max_left_ext - leftext] += cp->temp_in[i]; > + memcpy(out, cp->overlap, out_length * sizeof(*out)); > + memmove(cp->overlap, cp->overlap + out_length, > + (s->overlap_length - out_length) * sizeof(*cp->overlap)); > + memcpy(cp->overlap + s->overlap_length - out_length, cp->temp_in + leftext, > + out_length * sizeof(*cp->overlap)); > + } > + > + return 0; > +} > + > +static int next_pow2(int in) > +{ > + return 1 << (av_log2(in) + 1); > +} > + > +static void denoise_level(double *out, const double *in, > + const double *filter, > + double percent, int length) > +{ > + const double x = percent * 0.01; > + const double y = 1.0 - x; > + > + for (int i = 0; i < length; i++) > + out[i] = x * filter[i] + in[i] * y; > +} > + > +static double sqr(double in) > +{ > + return in * in; > +} > + > +static double measure_mean(const double *in, int length) > +{ > + double sum = 0.0; > + > + for (int i = 0; i < length; i++) > + sum += in[i]; > + > + return sum / length; > +} > + > +static double measure_absmean(const double *in, int length) > +{ > + double sum = 0.0; > + > + for (int i = 0; i < length; i++) > + sum += fabs(in[i]); > + > + return sum / length; > +} > + > +static double measure_stddev(const double *in, int length, double mean) > +{ > + double sum = 0.; > + > + for (int i = 0; i < length; i++) { > + sum += sqr(in[i] - mean); > + } > + > + return sqrt(sum / length); > +} > + > +static void noise_filter(const double stddev, const double *in, > + double *out, double absmean, double softness, > + double new_stddev, int length) > +{ > + for (int i = 0; i < length; i++) { > + if (new_stddev <= stddev) > + out[i] = 0.0; > + else if (fabs(in[i]) <= absmean) > + out[i] = 0.0; > + else > + out[i] = in[i] - FFSIGN(in[i]) * absmean / exp(3.0 * softness * (fabs(in[i]) - absmean) / absmean); > + } > +} > + > +typedef struct ThreadData { > + AVFrame *in, *out; > +} ThreadData; > + > +static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) > +{ > + AudioFWTDNContext *s = ctx->priv; > + ThreadData *td = arg; > + AVFrame *in = td->in; > + AVFrame *out = td->out; > + ChannelParams *cp = &s->cp[ch]; > + const double *src = (const double *)(in->extended_data[ch]); > + double *dst = (double *)out->extended_data[ch]; > + double *absmean = (double *)s->absmean->extended_data[ch]; > + double *new_absmean = (double *)s->new_absmean->extended_data[ch]; > + double *stddev = (double *)s->stddev->extended_data[ch]; > + double *new_stddev = (double *)s->new_stddev->extended_data[ch]; > + double *filter = (double *)s->filter->extended_data[ch]; > + double is_noise = 0.0; > + int ret; > + > + ret = forward(s, src, in->nb_samples, cp->output_coefs, cp->output_length, ch, s->sn); > + if (ret < 0) > + return ret; > + > + if (!s->got_profile && s->need_profile) { > + for (int level = 0; level <= s->levels; level++) { > + const int length = cp->output_length[level]; > + const double scale = sqrt(2.0 * log(length)); > + > + stddev[level] = measure_stddev(cp->output_coefs[level], length, > + measure_mean(cp->output_coefs[level], length)) * scale; > + absmean[level] = measure_absmean(cp->output_coefs[level], length) * scale; > + } > + } else if (!s->got_profile && !s->need_profile && !s->adaptive) { > + for (int level = 0; level <= s->levels; level++) { > + const int length = cp->output_length[level]; > + const double scale = sqrt(2.0 * log(length)); > + > + stddev[level] = 0.5 * s->sigma * scale; > + absmean[level] = 0.5 * s->sigma * scale; > + } > + } > + > + for (int level = 0; level <= s->levels; level++) { > + const int length = cp->output_length[level]; > + double vad; > + > + new_stddev[level] = measure_stddev(cp->output_coefs[level], length, > + measure_mean(cp->output_coefs[level], length)); > + new_absmean[level] = measure_absmean(cp->output_coefs[level], length); > + if (new_absmean[level] <= FLT_EPSILON) > + vad = 1.0; > + else > + vad = new_stddev[level] / new_absmean[level]; > + if (level < s->levels) > + is_noise += sqr(vad - 1.232); > + } > + > + is_noise *= in->sample_rate; > + is_noise /= s->nb_samples; > + for (int level = 0; level <= s->levels; level++) { > + const int length = cp->output_length[level]; > + const double scale = sqrt(2.0 * log(length)); > + > + if (is_noise < 0.05 && s->adaptive) { > + stddev[level] = new_stddev[level] * scale; > + absmean[level] = new_absmean[level] * scale; > + } > + > + noise_filter(stddev[level], cp->output_coefs[level], filter, absmean[level], > + s->softness, new_stddev[level], length); > + denoise_level(cp->filter_coefs[level], cp->output_coefs[level], filter, s->percent, length); > + } > + > + ret = inverse(s, cp->filter_coefs, cp->filter_length, dst, out->nb_samples, ch, s->sn); > + if (ret < 0) > + return ret; > + > + return 0; > +} > + > +static int filter_frame(AVFilterLink *inlink, AVFrame *in) > +{ > + AVFilterContext *ctx = inlink->dst; > + AudioFWTDNContext *s = ctx->priv; > + AVFilterLink *outlink = ctx->outputs[0]; > + ThreadData td; > + AVFrame *out; > + int eof = in == NULL; > + > + out = ff_get_audio_buffer(outlink, s->nb_samples); > + if (!out) { > + av_frame_free(&in); > + return AVERROR(ENOMEM); > + } > + if (in) { > + av_frame_copy_props(out, in); > + s->eof_pts = in->pts + in->nb_samples; > + } > + if (eof) > + out->pts = s->eof_pts - s->padd_samples; > + > + if (!in || in->nb_samples < s->nb_samples) { > + AVFrame *new_in = ff_get_audio_buffer(outlink, s->nb_samples); > + > + if (!new_in) { > + av_frame_free(&in); > + av_frame_free(&out); > + return AVERROR(ENOMEM); > + } > + if (in) > + av_frame_copy_props(new_in, in); > + > + s->padd_samples -= s->nb_samples - (in ? in->nb_samples: 0); > + if (in) > + av_samples_copy(new_in->extended_data, in->extended_data, 0, 0, > + in->nb_samples, in->channels, in->format); > + av_frame_free(&in); > + in = new_in; > + } > + > + td.in = in; > + td.out = out; > + ctx->internal->execute(ctx, s->filter_channel, &td, NULL, inlink->channels); > + if (s->need_profile) > + s->got_profile = 1; > + > + s->sn += s->nb_samples; > + > + if (s->drop_samples >= in->nb_samples) { > + s->drop_samples -= in->nb_samples; > + s->delay += in->nb_samples; > + av_frame_free(&in); > + av_frame_free(&out); > + FF_FILTER_FORWARD_STATUS(inlink, outlink); > + FF_FILTER_FORWARD_WANTED(outlink, inlink); > + return 0; > + } else if (s->drop_samples > 0) { > + for (int ch = 0; ch < out->channels; ch++) { > + memmove(out->extended_data[ch], > + out->extended_data[ch] + s->drop_samples * sizeof(double), > + (in->nb_samples - s->drop_samples) * sizeof(double)); > + } > + > + out->nb_samples = in->nb_samples - s->drop_samples; > + out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, outlink->sample_rate}, outlink->time_base); > + s->delay += s->drop_samples; > + s->drop_samples = 0; > + } else { > + if (s->padd_samples < 0 && eof) { > + out->nb_samples += s->padd_samples; > + s->padd_samples = 0; > + } > + if (!eof) > + out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, outlink->sample_rate}, outlink->time_base); > + } > + > + av_frame_free(&in); > + return ff_filter_frame(outlink, out); > +} > + > +static int max_left_ext(int wavelet_length, int levels) > +{ > + return (pow2(levels) - 1) * (wavelet_length - 1); > +} > + > +static int min_left_ext(int wavelet_length, int levels) > +{ > + return (pow2(levels) - 1) * (wavelet_length - 2); > +} > + > +static int config_output(AVFilterLink *outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + AudioFWTDNContext *s = ctx->priv; > + > + switch (s->wavelet_type) { > + case SYM2: > + s->wavelet_length = 4; > + s->lp = sym2_lp; > + s->hp = sym2_hp; > + s->ilp = sym2_ilp; > + s->ihp = sym2_ihp; > + break; > + case SYM4: > + s->wavelet_length = 8; > + s->lp = sym4_lp; > + s->hp = sym4_hp; > + s->ilp = sym4_ilp; > + s->ihp = sym4_ihp; > + break; > + case RBIOR68: > + s->wavelet_length = 18; > + s->lp = rbior68_lp; > + s->hp = rbior68_hp; > + s->ilp = rbior68_ilp; > + s->ihp = rbior68_ihp; > + break; > + case DEB10: > + s->wavelet_length = 20; > + s->lp = deb10_lp; > + s->hp = deb10_hp; > + s->ilp = deb10_ilp; > + s->ihp = deb10_ihp; > + case SYM10: > + s->wavelet_length = 20; > + s->lp = sym10_lp; > + s->hp = sym10_hp; > + s->ilp = sym10_ilp; > + s->ihp = sym10_ihp; > + break; > + case COIF5: > + s->wavelet_length = 30; > + s->lp = coif5_lp; > + s->hp = coif5_hp; > + s->ilp = coif5_ilp; > + s->ihp = coif5_ihp; > + break; > + case BL3: > + s->wavelet_length = 42; > + s->lp = bl3_lp; > + s->hp = bl3_hp; > + s->ilp = bl3_ilp; > + s->ihp = bl3_ihp; > + break; > + default: > + av_assert0(0); > + } > + > + s->levels = FFMIN(s->levels, lrint(log(s->nb_samples / (s->wavelet_length - 1.0)) / M_LN2)); > + av_log(ctx, AV_LOG_VERBOSE, "levels: %d\n", s->levels); > + s->filter_channel = filter_channel; > + > + s->stddev = ff_get_audio_buffer(outlink, MAX_LEVELS); > + s->new_stddev = ff_get_audio_buffer(outlink, MAX_LEVELS); > + s->filter = ff_get_audio_buffer(outlink, s->nb_samples); > + s->absmean = ff_get_audio_buffer(outlink, MAX_LEVELS); > + s->new_absmean = ff_get_audio_buffer(outlink, MAX_LEVELS); > + if (!s->stddev || !s->absmean || !s->filter || > + !s->new_stddev || !s->new_absmean) > + return AVERROR(ENOMEM); > + > + s->channels = outlink->channels; > + s->overlap_length = max_left_ext(s->wavelet_length, s->levels); > + s->prev_length = s->overlap_length; > + s->drop_samples = s->overlap_length; > + s->padd_samples = s->overlap_length; > + s->sn = 1; > + > + s->cp = av_calloc(s->channels, sizeof(*s->cp)); > + if (!s->cp) > + return AVERROR(ENOMEM); > + > + for (int ch = 0; ch < s->channels; ch++) { > + ChannelParams *cp = &s->cp[ch]; > + > + cp->output_coefs = av_calloc(s->levels + 1, sizeof(*cp->output_coefs)); > + cp->filter_coefs = av_calloc(s->levels + 1, sizeof(*cp->filter_coefs)); > + cp->output_length = av_calloc(s->levels + 1, sizeof(*cp->output_length)); > + cp->filter_length = av_calloc(s->levels + 1, sizeof(*cp->filter_length)); > + cp->buffer_length = next_pow2(s->wavelet_length); > + cp->buffer = av_calloc(cp->buffer_length, sizeof(*cp->buffer)); > + cp->buffer2 = av_calloc(cp->buffer_length, sizeof(*cp->buffer2)); > + cp->subbands_to_free = av_calloc(s->levels + 1, sizeof(*cp->subbands_to_free)); > + cp->prev = av_calloc(s->prev_length, sizeof(*cp->prev)); > + cp->overlap = av_calloc(s->overlap_length, sizeof(*cp->overlap)); > + cp->max_left_ext = max_left_ext(s->wavelet_length, s->levels); > + cp->min_left_ext = min_left_ext(s->wavelet_length, s->levels); > + if (!cp->output_coefs || !cp->filter_coefs || !cp->output_length || > + !cp->filter_length || !cp->subbands_to_free || !cp->prev || !cp->overlap || > + !cp->buffer || !cp->buffer2) > + return AVERROR(ENOMEM); > + } > + > + return 0; > +} > + > +static int activate(AVFilterContext *ctx) > +{ > + AVFilterLink *inlink = ctx->inputs[0]; > + AVFilterLink *outlink = ctx->outputs[0]; > + AudioFWTDNContext *s = ctx->priv; > + AVFrame *in = NULL; > + int ret, status; > + int64_t pts; > + > + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); > + > + ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in); > + if (ret < 0) > + return ret; > + if (ret > 0) > + return filter_frame(inlink, in); > + > + if (ff_inlink_acknowledge_status(inlink, &status, &pts)) { > + if (status == AVERROR_EOF) { > + while (s->padd_samples != 0) { > + ret = filter_frame(inlink, NULL); > + if (ret < 0) > + return ret; > + } > + ff_outlink_set_status(outlink, status, pts); > + return ret; > + } > + } > + FF_FILTER_FORWARD_WANTED(outlink, inlink); > + > + return FFERROR_NOT_READY; > +} > + > +static av_cold void uninit(AVFilterContext *ctx) > +{ > + AudioFWTDNContext *s = ctx->priv; > + > + av_frame_free(&s->filter); > + av_frame_free(&s->new_stddev); > + av_frame_free(&s->stddev); > + av_frame_free(&s->new_absmean); > + av_frame_free(&s->absmean); > + > + for (int ch = 0; s->cp && ch < s->channels; ch++) { > + ChannelParams *cp = &s->cp[ch]; > + > + av_freep(&cp->tempa); > + av_freep(&cp->tempd); > + av_freep(&cp->temp_in); > + av_freep(&cp->buffer); > + av_freep(&cp->buffer2); > + av_freep(&cp->prev); > + av_freep(&cp->overlap); > + > + av_freep(&cp->output_length); > + av_freep(&cp->filter_length); > + > + if (cp->output_coefs) { > + for (int level = 0; level <= s->levels; level++) > + av_freep(&cp->output_coefs[level]); > + } > + > + if (cp->subbands_to_free) { > + for (int level = 0; level <= s->levels; level++) > + av_freep(&cp->subbands_to_free[level]); > + } > + > + av_freep(&cp->subbands_to_free); > + av_freep(&cp->output_coefs); > + av_freep(&cp->filter_coefs); > + } > + > + av_freep(&s->cp); > +} > + > +static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, > + char *res, int res_len, int flags) > +{ > + AudioFWTDNContext *s = ctx->priv; > + int ret; > + > + ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); > + if (ret < 0) > + return ret; > + > + if (!strcmp(cmd, "profile") && s->need_profile) > + s->got_profile = 0; > + > + return 0; > +} > + > +static const AVFilterPad inputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + }, > + { NULL } > +}; > + > +static const AVFilterPad outputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + .config_props = config_output, > + }, > + { NULL } > +}; > + > +AVFilter ff_af_afwtdn = { > + .name = "afwtdn", > + .description = NULL_IF_CONFIG_SMALL("Denoise audio stream using Wavelets."), > + .query_formats = query_formats, > + .priv_size = sizeof(AudioFWTDNContext), > + .priv_class = &afwtdn_class, > + .activate = activate, > + .uninit = uninit, > + .inputs = inputs, > + .outputs = outputs, > + .process_command = process_command, > + .flags = AVFILTER_FLAG_SLICE_THREADS, > +}; > diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c > index 1183e40267..de5884529c 100644 > --- a/libavfilter/allfilters.c > +++ b/libavfilter/allfilters.c > @@ -43,6 +43,7 @@ extern AVFilter ff_af_afftdn; > extern AVFilter ff_af_afftfilt; > extern AVFilter ff_af_afir; > extern AVFilter ff_af_aformat; > +extern AVFilter ff_af_afwtdn; > extern AVFilter ff_af_agate; > extern AVFilter ff_af_aiir; > extern AVFilter ff_af_aintegral;
On 6/11/20, Nicolas George <george@nsup.org> wrote: > Paul B Mahol (12020-06-11): >> Signed-off-by: Paul B Mahol <onemda@gmail.com> >> --- >> doc/filters.texi | 60 ++ >> libavfilter/Makefile | 1 + >> libavfilter/af_afwtdn.c | 1345 ++++++++++++++++++++++++++++++++++++++ >> libavfilter/allfilters.c | 1 + >> 4 files changed, 1407 insertions(+) >> create mode 100644 libavfilter/af_afwtdn.c > > I still oppose to this filter on the basis of the name. > >> >> diff --git a/doc/filters.texi b/doc/filters.texi >> index c2960e33c7..d89ebc5122 100644 >> --- a/doc/filters.texi >> +++ b/doc/filters.texi >> @@ -1314,6 +1314,66 @@ Force the output to either unsigned 8-bit or signed >> 16-bit stereo >> aformat=sample_fmts=u8|s16:channel_layouts=stereo >> @end example >> > >> +@section afwtdn >> +Reduce broadband noise from input samples using Wavelets. > > Please document in user-oriented terms and not in developer-oriented the > benefits of this filter for the user. You need to explain WHAT it does > different from the n other denoisers, not HOW it does it. > >> + >> +A description of the accepted options follows. >> + >> +@table @option >> +@item sigma >> +Set the noise sigma, allowed range is from 0 to 1. >> +Default value is 0. >> +This option controls strength of denoising applied to input samples. >> +Most useful way to set this option is via decibels, eg. -45dB. >> + >> +@item levels >> +Set the number of wavelet levels of decomposition. >> +Allowed range is from 1 to 12. >> +Default value is 10. >> +Setting this too low make denoising performance very poor. >> + >> +@item wavet >> +Set wavelet type for decomposition of input frame. >> +They are sorted by number of coefficients, from lowest to highest. >> +More coefficients means worse filtering speed, but overall better >> quality. >> +Available wavelets are: >> + >> +@table @samp >> +@item sym2 >> +@item sym4 >> +@item rbior68 >> +@item deb10 >> +@item sym10 >> +@item coif5 >> +@item bl3 >> +@end table >> + >> +@item percent >> +Set percent of full denoising. Allowed range is from 0 to 100 percent. >> +Default value is 85 percent or partial denoising. >> + >> +@item profile >> +If enabled, first input frame will be used as noise profile. >> +If first frame samples contain non-noise performance will be very poor. >> + >> +@item adaptive >> +If enabled, input frames are analyzed for presence of noise. >> +If noise is detected with high possibility then input frame profile will >> be >> +used for processing following frames, until new noise frame is detected. >> + >> +@item samples >> +Set size of single frame in number of samples. Allowed range is from 512 >> to >> +65536. Default frame size is 8192 samples. >> + >> +@item softness >> +Set softness applied inside thresholding function. Allowed range is from >> 0 to >> +10. Default softness is 1. >> +@end table >> + >> +@subsection Commands >> + >> +This filter supports subset of its options as @ref{commands}. >> + >> @section agate >> >> A gate is mainly used to reduce lower parts of a signal. This kind of >> signal >> diff --git a/libavfilter/Makefile b/libavfilter/Makefile >> index 5123540653..191826a622 100644 >> --- a/libavfilter/Makefile >> +++ b/libavfilter/Makefile >> @@ -50,6 +50,7 @@ OBJS-$(CONFIG_AFFTDN_FILTER) += >> af_afftdn.o >> OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o >> OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o >> OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o >> +OBJS-$(CONFIG_AFWTDN_FILTER) += af_afwtdn.o >> OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o >> OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o >> OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o >> diff --git a/libavfilter/af_afwtdn.c b/libavfilter/af_afwtdn.c >> new file mode 100644 >> index 0000000000..d2793d4d92 >> --- /dev/null >> +++ b/libavfilter/af_afwtdn.c >> @@ -0,0 +1,1345 @@ >> +/* >> + * Copyright (c) 2020 Paul B Mahol >> + * >> + * This file is part of FFmpeg. >> + * >> + * FFmpeg is free software; you can redistribute it and/or >> + * modify it under the terms of the GNU Lesser General Public >> + * License as published by the Free Software Foundation; either >> + * version 2.1 of the License, or (at your option) any later version. >> + * >> + * FFmpeg is distributed in the hope that it will be useful, >> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >> + * Lesser General Public License for more details. >> + * >> + * You should have received a copy of the GNU Lesser General Public >> + * License along with FFmpeg; if not, write to the Free Software >> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA >> 02110-1301 USA >> + */ >> + >> +#include <float.h> >> + >> +#include "libavutil/avassert.h" >> +#include "libavutil/avstring.h" >> +#include "libavutil/opt.h" >> +#include "avfilter.h" >> +#include "audio.h" >> +#include "filters.h" >> +#include "formats.h" >> + >> +enum WaveletTypes { >> + SYM2, >> + SYM4, >> + RBIOR68, >> + DEB10, >> + SYM10, >> + COIF5, >> + BL3, >> + NB_WAVELET_TYPES, >> +}; >> + > >> +static const double bl3_lp[42] = { >> + 0.000146098, -0.000232304, -0.000285414, 0.000462093, 0.000559952, >> + -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426, >> + -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331, >> + 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147, >> + 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869, >> + 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477, >> + -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748, >> + -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304, >> + 0.000146098, 0.0, >> +}; >> + >> +static const double bl3_hp[42] = { >> + 0.0, 0.000146098, 0.000232304, -0.000285414, -0.000462093, >> 0.000559952, >> + 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426, >> + -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331, >> + -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147, >> + 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869, >> + -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477, >> + -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748, >> + 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304, >> + 0.000146098, >> +}; >> + >> +static const double bl3_ilp[42] = { >> + 0.0, 0.000146098, -0.000232304, -0.000285414, 0.000462093, >> 0.000559952, >> + -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426, >> + -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331, >> + 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147, >> + 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869, >> + 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477, >> + -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748, >> + -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304, >> + 0.000146098, >> +}; >> + >> +static const double bl3_ihp[42] = { >> + 0.000146098, 0.000232304, -0.000285414, -0.000462093, 0.000559952, >> + 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426, >> + -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331, >> + -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147, >> + 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869, >> + -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477, >> + -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748, >> + 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304, >> + 0.000146098, >> +}; >> + >> +static const double sym10_lp[20] = { >> + 0.0007701598091144901, 9.563267072289475e-05, >> + -0.008641299277022422, -0.0014653825813050513, >> + 0.0459272392310922, 0.011609893903711381, >> + -0.15949427888491757, -0.07088053578324385, >> + 0.47169066693843925, 0.7695100370211071, >> + 0.38382676106708546, -0.03553674047381755, >> + -0.0319900568824278, 0.04999497207737669, >> + 0.005764912033581909, -0.02035493981231129, >> + -0.0008043589320165449, 0.004593173585311828, >> + 5.7036083618494284e-05, -0.0004593294210046588, >> +}; >> + >> +static const double sym10_hp[20] = { >> + 0.0004593294210046588, 5.7036083618494284e-05, >> + -0.004593173585311828, -0.0008043589320165449, >> + 0.02035493981231129, 0.005764912033581909, >> + -0.04999497207737669, -0.0319900568824278, >> + 0.03553674047381755, 0.38382676106708546, >> + -0.7695100370211071, 0.47169066693843925, >> + 0.07088053578324385, -0.15949427888491757, >> + -0.011609893903711381, 0.0459272392310922, >> + 0.0014653825813050513, -0.008641299277022422, >> + -9.563267072289475e-05, 0.0007701598091144901, >> +}; >> + >> +static const double sym10_ilp[20] = { >> + -0.0004593294210046588, 5.7036083618494284e-05, >> + 0.004593173585311828, -0.0008043589320165449, >> + -0.02035493981231129, 0.005764912033581909, >> + 0.04999497207737669, -0.0319900568824278, >> + -0.03553674047381755, 0.38382676106708546, >> + 0.7695100370211071, 0.47169066693843925, >> + -0.07088053578324385, -0.15949427888491757, >> + 0.011609893903711381, 0.0459272392310922, >> + -0.0014653825813050513, -0.008641299277022422, >> + 9.563267072289475e-05, 0.0007701598091144901, >> +}; >> + >> +static const double sym10_ihp[20] = { >> + 0.0007701598091144901, -9.563267072289475e-05, >> + -0.008641299277022422, 0.0014653825813050513, >> + 0.0459272392310922, -0.011609893903711381, >> + -0.15949427888491757, 0.07088053578324385, >> + 0.47169066693843925, -0.7695100370211071, >> + 0.38382676106708546, 0.03553674047381755, >> + -0.0319900568824278, -0.04999497207737669, >> + 0.005764912033581909, 0.02035493981231129, >> + -0.0008043589320165449, -0.004593173585311828, >> + 5.7036083618494284e-05, 0.0004593294210046588, >> +}; >> + >> +static const double rbior68_lp[18] = { >> + 0.0, 0.0, 0.0, 0.0, >> + 0.014426282505624435, 0.014467504896790148, >> + -0.07872200106262882, -0.04036797903033992, >> + 0.41784910915027457, 0.7589077294536541, >> + 0.41784910915027457, -0.04036797903033992, >> + -0.07872200106262882, 0.014467504896790148, >> + 0.014426282505624435, 0.0, 0.0, 0.0, >> +}; >> + >> +static const double rbior68_hp[18] = { >> + -0.0019088317364812906, -0.0019142861290887667, >> + 0.016990639867602342, 0.01193456527972926, >> + -0.04973290349094079, -0.07726317316720414, >> + 0.09405920349573646, 0.4207962846098268, >> + -0.8259229974584023, 0.4207962846098268, >> + 0.09405920349573646, -0.07726317316720414, >> + -0.04973290349094079, 0.01193456527972926, >> + 0.016990639867602342, -0.0019142861290887667, >> + -0.0019088317364812906, 0.0, >> +}; >> + >> +static const double rbior68_ilp[18] = { >> + 0.0019088317364812906, -0.0019142861290887667, >> + -0.016990639867602342, 0.01193456527972926, >> + 0.04973290349094079, -0.07726317316720414, >> + -0.09405920349573646, 0.4207962846098268, >> + 0.8259229974584023, 0.4207962846098268, >> + -0.09405920349573646, -0.07726317316720414, >> + 0.04973290349094079, 0.01193456527972926, >> + -0.016990639867602342, -0.0019142861290887667, >> + 0.0019088317364812906, 0.0, >> +}; >> + >> +static const double rbior68_ihp[18] = { >> + 0.0, 0.0, 0.0, 0.0, >> + 0.014426282505624435, -0.014467504896790148, >> + -0.07872200106262882, 0.04036797903033992, >> + 0.41784910915027457, -0.7589077294536541, >> + 0.41784910915027457, 0.04036797903033992, >> + -0.07872200106262882, -0.014467504896790148, >> + 0.014426282505624435, 0.0, 0.0, 0.0, >> +}; >> + >> +static const double coif5_lp[30] = { >> + -9.517657273819165e-08, -1.6744288576823017e-07, >> + 2.0637618513646814e-06, 3.7346551751414047e-06, >> + -2.1315026809955787e-05, -4.134043227251251e-05, >> + 0.00014054114970203437, 0.00030225958181306315, >> + -0.0006381313430451114, -0.0016628637020130838, >> + 0.0024333732126576722, 0.006764185448053083, >> + -0.009164231162481846, -0.01976177894257264, >> + 0.03268357426711183, 0.0412892087501817, >> + -0.10557420870333893, -0.06203596396290357, >> + 0.4379916261718371, 0.7742896036529562, >> + 0.4215662066908515, -0.05204316317624377, >> + -0.09192001055969624, 0.02816802897093635, >> + 0.023408156785839195, -0.010131117519849788, >> + -0.004159358781386048, 0.0021782363581090178, >> + 0.00035858968789573785, -0.00021208083980379827, >> +}; >> + >> +static const double coif5_hp[30] = { >> + 0.00021208083980379827, 0.00035858968789573785, >> + -0.0021782363581090178, -0.004159358781386048, >> + 0.010131117519849788, 0.023408156785839195, >> + -0.02816802897093635, -0.09192001055969624, >> + 0.05204316317624377, 0.4215662066908515, >> + -0.7742896036529562, 0.4379916261718371, >> + 0.06203596396290357, -0.10557420870333893, >> + -0.0412892087501817, 0.03268357426711183, >> + 0.01976177894257264, -0.009164231162481846, >> + -0.006764185448053083, 0.0024333732126576722, >> + 0.0016628637020130838, -0.0006381313430451114, >> + -0.00030225958181306315, 0.00014054114970203437, >> + 4.134043227251251e-05, -2.1315026809955787e-05, >> + -3.7346551751414047e-06, 2.0637618513646814e-06, >> + 1.6744288576823017e-07, -9.517657273819165e-08, >> +}; >> + >> +static const double coif5_ilp[30] = { >> + -0.00021208083980379827, 0.00035858968789573785, >> + 0.0021782363581090178, -0.004159358781386048, >> + -0.010131117519849788, 0.023408156785839195, >> + 0.02816802897093635, -0.09192001055969624, >> + -0.05204316317624377, 0.4215662066908515, >> + 0.7742896036529562, 0.4379916261718371, >> + -0.06203596396290357, -0.10557420870333893, >> + 0.0412892087501817, 0.03268357426711183, >> + -0.01976177894257264, -0.009164231162481846, >> + 0.006764185448053083, 0.0024333732126576722, >> + -0.0016628637020130838, -0.0006381313430451114, >> + 0.00030225958181306315, 0.00014054114970203437, >> + -4.134043227251251e-05, -2.1315026809955787e-05, >> + 3.7346551751414047e-06, 2.0637618513646814e-06, >> + -1.6744288576823017e-07, -9.517657273819165e-08, >> +}; >> + >> +static const double coif5_ihp[30] = { >> + -9.517657273819165e-08, 1.6744288576823017e-07, >> + 2.0637618513646814e-06, -3.7346551751414047e-06, >> + -2.1315026809955787e-05, 4.134043227251251e-05, >> + 0.00014054114970203437, -0.00030225958181306315, >> + -0.0006381313430451114, 0.0016628637020130838, >> + 0.0024333732126576722, -0.006764185448053083, >> + -0.009164231162481846, 0.01976177894257264, >> + 0.03268357426711183, -0.0412892087501817, >> + -0.10557420870333893, 0.06203596396290357, >> + 0.4379916261718371, -0.7742896036529562, >> + 0.4215662066908515, 0.05204316317624377, >> + -0.09192001055969624, -0.02816802897093635, >> + 0.023408156785839195, 0.010131117519849788, >> + -0.004159358781386048, -0.0021782363581090178, >> + 0.00035858968789573785, 0.00021208083980379827, >> +}; >> + >> +static const double deb10_lp[20] = { >> + -1.326420300235487e-05, 9.358867000108985e-05, >> + -0.0001164668549943862, -0.0006858566950046825, >> + 0.00199240529499085, 0.0013953517469940798, >> + -0.010733175482979604, 0.0036065535669883944, >> + 0.03321267405893324, -0.02945753682194567, >> + -0.07139414716586077, 0.09305736460380659, >> + 0.12736934033574265, -0.19594627437659665, >> + -0.24984642432648865, 0.2811723436604265, >> + 0.6884590394525921, 0.5272011889309198, >> + 0.18817680007762133, 0.026670057900950818, >> +}; >> + >> +static const double deb10_hp[20] = { >> + -0.026670057900950818, 0.18817680007762133, >> + -0.5272011889309198, 0.6884590394525921, >> + -0.2811723436604265, -0.24984642432648865, >> + 0.19594627437659665, 0.12736934033574265, >> + -0.09305736460380659, -0.07139414716586077, >> + 0.02945753682194567, 0.03321267405893324, >> + -0.0036065535669883944, -0.010733175482979604, >> + -0.0013953517469940798, 0.00199240529499085, >> + 0.0006858566950046825, -0.0001164668549943862, >> + -9.358867000108985e-05, -1.326420300235487e-05, >> +}; >> + >> +static const double deb10_ilp[20] = { >> + 0.026670057900950818, 0.18817680007762133, >> + 0.5272011889309198, 0.6884590394525921, >> + 0.2811723436604265, -0.24984642432648865, >> + -0.19594627437659665, 0.12736934033574265, >> + 0.09305736460380659, -0.07139414716586077, >> + -0.02945753682194567, 0.03321267405893324, >> + 0.0036065535669883944, -0.010733175482979604, >> + 0.0013953517469940798, 0.00199240529499085, >> + -0.0006858566950046825, -0.0001164668549943862, >> + 9.358867000108985e-05, -1.326420300235487e-05, >> +}; >> + >> +static const double deb10_ihp[20] = { >> + -1.326420300235487e-05, -9.358867000108985e-05, >> + -0.0001164668549943862, 0.0006858566950046825, >> + 0.00199240529499085, -0.0013953517469940798, >> + -0.010733175482979604, -0.0036065535669883944, >> + 0.03321267405893324, 0.02945753682194567, >> + -0.07139414716586077, -0.09305736460380659, >> + 0.12736934033574265, 0.19594627437659665, >> + -0.24984642432648865, -0.2811723436604265, >> + 0.6884590394525921, -0.5272011889309198, >> + 0.18817680007762133, -0.026670057900950818, >> +}; >> + >> +static const double sym4_lp[8] = { >> + -0.07576571478927333, >> + -0.02963552764599851, >> + 0.49761866763201545, >> + 0.8037387518059161, >> + 0.29785779560527736, >> + -0.09921954357684722, >> + -0.012603967262037833, >> + 0.0322231006040427, >> +}; >> + >> +static const double sym4_hp[8] = { >> + -0.0322231006040427, >> + -0.012603967262037833, >> + 0.09921954357684722, >> + 0.29785779560527736, >> + -0.8037387518059161, >> + 0.49761866763201545, >> + 0.02963552764599851, >> + -0.07576571478927333, >> +}; >> + >> +static const double sym4_ilp[8] = { >> + 0.0322231006040427, >> + -0.012603967262037833, >> + -0.09921954357684722, >> + 0.29785779560527736, >> + 0.8037387518059161, >> + 0.49761866763201545, >> + -0.02963552764599851, >> + -0.07576571478927333, >> +}; >> + >> +static const double sym4_ihp[8] = { >> + -0.07576571478927333, >> + 0.02963552764599851, >> + 0.49761866763201545, >> + -0.8037387518059161, >> + 0.29785779560527736, >> + 0.09921954357684722, >> + -0.012603967262037833, >> + -0.0322231006040427, >> +}; >> + >> +static const double sym2_lp[4] = { >> + -0.12940952255092145, 0.22414386804185735, >> + 0.836516303737469, 0.48296291314469025, >> +}; >> + >> +static const double sym2_hp[4] = { >> + -0.48296291314469025, 0.836516303737469, >> + -0.22414386804185735, -0.12940952255092145, >> +}; >> + >> +static const double sym2_ilp[4] = { >> + 0.48296291314469025, 0.836516303737469, >> + 0.22414386804185735, -0.12940952255092145, >> +}; >> + >> +static const double sym2_ihp[4] = { >> + -0.12940952255092145, -0.22414386804185735, >> + 0.836516303737469, -0.48296291314469025, >> +}; > > You did not compute these numbers in your head nor did you type them by > hand: they are not source code. We must include the whole source code. > This is part of basic math, you seems to lack basic math knowledge. >> + >> +#define MAX_LEVELS 13 >> + >> +typedef struct ChannelParams { >> + int *output_length; >> + int *filter_length; >> + double **output_coefs; >> + double **subbands_to_free; >> + double **filter_coefs; >> + >> + int tempa_length; >> + int tempa_len_max; >> + int temp_in_length; >> + int temp_in_max_length; >> + int buffer_length; >> + int min_left_ext; >> + int max_left_ext; >> + >> + double *tempa; >> + double *tempd; >> + double *temp_in; >> + double *buffer; >> + double *buffer2; >> + double *prev; >> + double *overlap; >> +} ChannelParams; >> + >> +typedef struct AudioFWTDNContext { >> + const AVClass *class; >> + >> + double sigma; >> + double percent; >> + double softness; >> + >> + uint64_t sn; >> + int64_t eof_pts; >> + >> + int wavelet_type; >> + int channels; >> + int nb_samples; >> + int levels; >> + int wavelet_length; >> + int need_profile; >> + int got_profile; >> + int adaptive; >> + >> + int delay; >> + int drop_samples; >> + int padd_samples; >> + int overlap_length; >> + int prev_length; >> + ChannelParams *cp; >> + >> + const double *lp, *hp; >> + const double *ilp, *ihp; >> + >> + AVFrame *stddev, *absmean, *filter; >> + AVFrame *new_stddev, *new_absmean; >> + >> + int (*filter_channel)(AVFilterContext *ctx, void *arg, int ch, int >> nb_jobs); >> +} AudioFWTDNContext; >> + >> +#define OFFSET(x) offsetof(AudioFWTDNContext, x) >> +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM >> +#define AFR >> AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM >> + >> +static const AVOption afwtdn_options[] = { >> + { "sigma", "set noise sigma", OFFSET(sigma), AV_OPT_TYPE_DOUBLE, >> {.dbl=0}, 0, 1, AFR }, >> + { "levels", "set number of wavelet levels", OFFSET(levels), >> AV_OPT_TYPE_INT, {.i64=10}, 1, MAX_LEVELS-1, AF }, >> + { "wavet", "set wavelet type", OFFSET(wavelet_type), AV_OPT_TYPE_INT, >> {.i64=SYM10}, 0, NB_WAVELET_TYPES - 1, AF, "wavet" }, >> + { "sym2", "sym2", 0, AV_OPT_TYPE_CONST, {.i64=SYM2}, 0, 0, AF, >> "wavet" }, >> + { "sym4", "sym4", 0, AV_OPT_TYPE_CONST, {.i64=SYM4}, 0, 0, AF, >> "wavet" }, >> + { "rbior68", "rbior68", 0, AV_OPT_TYPE_CONST, {.i64=RBIOR68}, 0, 0, >> AF, "wavet" }, >> + { "deb10", "deb10", 0, AV_OPT_TYPE_CONST, {.i64=DEB10}, 0, 0, AF, >> "wavet" }, >> + { "sym10", "sym10", 0, AV_OPT_TYPE_CONST, {.i64=SYM10}, 0, 0, AF, >> "wavet" }, >> + { "coif5", "coif5", 0, AV_OPT_TYPE_CONST, {.i64=COIF5}, 0, 0, AF, >> "wavet" }, >> + { "bl3", "bl3", 0, AV_OPT_TYPE_CONST, {.i64=BL3}, 0, 0, AF, "wavet" >> }, >> + { "percent", "set percent of full denoising", >> OFFSET(percent),AV_OPT_TYPE_DOUBLE, {.dbl=85}, 0, 100, AFR }, >> + { "profile", "profile noise", OFFSET(need_profile), AV_OPT_TYPE_BOOL, >> {.i64=0}, 0, 1, AFR }, >> + { "adaptive", "adaptive profiling of noise", OFFSET(adaptive), >> AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AFR }, >> + { "samples", "set frame size in number of samples", >> OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=8192}, 512, 65536, AF }, >> + { "softness", "set thresholding softness", OFFSET(softness), >> AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 10, AFR }, >> + { NULL } >> +}; >> + >> +AVFILTER_DEFINE_CLASS(afwtdn); >> + >> +static int query_formats(AVFilterContext *ctx) >> +{ >> + AVFilterFormats *formats = NULL; >> + AVFilterChannelLayouts *layouts = NULL; >> + static const enum AVSampleFormat sample_fmts[] = { >> + AV_SAMPLE_FMT_DBLP, >> + AV_SAMPLE_FMT_NONE >> + }; >> + int ret; >> + >> + formats = ff_make_format_list(sample_fmts); >> + if (!formats) >> + return AVERROR(ENOMEM); >> + ret = ff_set_common_formats(ctx, formats); >> + if (ret < 0) >> + return ret; >> + >> + layouts = ff_all_channel_counts(); >> + if (!layouts) >> + return AVERROR(ENOMEM); >> + >> + ret = ff_set_common_channel_layouts(ctx, layouts); >> + if (ret < 0) >> + return ret; >> + >> + formats = ff_all_samplerates(); >> + return ff_set_common_samplerates(ctx, formats); >> +} >> + >> +#define pow2(x) (1U << (x)) >> +#define mod_pow2(x, power_of_two) ((x) & ((power_of_two) - 1)) >> + >> +static void conv_down(double *in, int in_length, double *low, double >> *high, >> + int out_length, const double *lp, const double >> *hp, >> + int wavelet_length, int skip, >> + double *buffer, int buffer_length) >> +{ >> + double thigh = 0.0, tlow = 0.0; >> + int buff_idx = 1 + skip; >> + >> + memcpy(buffer, in, buff_idx * sizeof(*buffer)); >> + memset(buffer + buff_idx, 0, (buffer_length - buff_idx) * >> sizeof(*buffer)); >> + >> + for (int i = 0; i < out_length - 1; i++) { >> + double thigh = 0.0, tlow = 0.0; >> + >> + for (int j = 0; j < wavelet_length; j++) { >> + const int idx = mod_pow2(-j + buff_idx - 1, buffer_length); >> + const double btemp = buffer[idx]; >> + >> + thigh += btemp * hp[j]; >> + tlow += btemp * lp[j]; >> + } >> + >> + high[i] = thigh; >> + low[i] = tlow; >> + buffer[buff_idx++] = in[2 * i + 1 + skip]; >> + buffer[buff_idx++] = in[2 * i + 2 + skip]; >> + buff_idx = mod_pow2(buff_idx, buffer_length); >> + } >> + >> + for (int i = 0; i < wavelet_length; i++) { >> + const int idx = mod_pow2(-i + buff_idx - 1, buffer_length); >> + const double btemp = buffer[idx]; >> + >> + thigh += btemp * hp[i]; >> + tlow += btemp * lp[i]; >> + } >> + >> + high[out_length - 1] = thigh; >> + low[out_length - 1] = tlow; >> +} >> + >> +static int left_ext(int wavelet_length, int levels, uint64_t sn) >> +{ >> + if (!sn) >> + return 0; >> + return (pow2(levels) - 1) * (wavelet_length - 2) + mod_pow2(sn, >> pow2(levels)); >> +} >> + >> +static int nb_coefs(int length, int level, uint64_t sn) >> +{ >> + const int pow2_level = pow2(level); >> + >> + return (sn + length) / pow2_level - sn / pow2_level; >> +} >> + >> +static int reallocate_inputs(double **out, int *out_length, >> + int in_length, int levels, int ch, uint64_t >> sn) >> +{ >> + const int temp_length = nb_coefs(in_length, levels, sn); >> + >> + for (int level = 0; level < levels; level++) { >> + const int temp_length = nb_coefs(in_length, level + 1, sn); >> + >> + if (temp_length > out_length[level]) { >> + av_freep(&out[level]); >> + out_length[level] = 0; >> + >> + out[level] = av_calloc(temp_length + 1, sizeof(**out)); >> + if (!out[level]) >> + return AVERROR(ENOMEM); >> + out_length[level] = temp_length + 1; >> + } >> + >> + memset(out[level] + temp_length, 0, >> + (out_length[level] - temp_length) * sizeof(**out)); >> + out_length[level] = temp_length; >> + } >> + >> + if (temp_length > out_length[levels]) { >> + av_freep(&out[levels]); >> + out_length[levels] = 0; >> + >> + out[levels] = av_calloc(temp_length + 1, sizeof(**out)); >> + if (!out[levels]) >> + return AVERROR(ENOMEM); >> + out_length[levels] = temp_length + 1; >> + } >> + >> + memset(out[levels] + temp_length, 0, >> + (out_length[levels] - temp_length) * sizeof(**out)); >> + out_length[levels] = temp_length; >> + >> + return 0; >> +} >> + >> +static int max_left_zeros_inverse(int levels, int level, int >> wavelet_length) >> +{ >> + return (pow2(levels - level) - 1) * (wavelet_length - 1); >> +} >> + >> +static int reallocate_outputs(AudioFWTDNContext *s, >> + double **out, int *out_length, >> + int in_length, int levels, int ch, uint64_t >> sn) >> +{ >> + ChannelParams *cp = &s->cp[ch]; >> + int temp_length = 0; >> + int add = 0; >> + >> + for (int level = 0; level < levels; level++) { >> + temp_length = nb_coefs(in_length, level + 1, sn); >> + if (temp_length > out_length[level]) { >> + av_freep(&cp->subbands_to_free[level]); >> + out_length[level] = 0; >> + >> + add = max_left_zeros_inverse(levels, level + 1, >> s->wavelet_length); >> + cp->subbands_to_free[level] = av_calloc(add + temp_length + >> 1, sizeof(**out)); >> + if (!cp->subbands_to_free[level]) >> + return AVERROR(ENOMEM); >> + out_length[level] = add + temp_length + 1; >> + out[level] = cp->subbands_to_free[level] + add; >> + } >> + >> + memset(out[level] + temp_length, 0, >> + FFMAX(out_length[level] - temp_length - add, 0) * >> sizeof(**out)); >> + out_length[level] = temp_length; >> + } >> + >> + temp_length = nb_coefs(in_length, levels, sn); >> + if (temp_length > out_length[levels]) { >> + av_freep(&cp->subbands_to_free[levels]); >> + out_length[levels] = 0; >> + >> + cp->subbands_to_free[levels] = av_calloc(temp_length + 1, >> sizeof(**out)); >> + if (!cp->subbands_to_free[levels]) >> + return AVERROR(ENOMEM); >> + out_length[levels] = temp_length + 1; >> + out[levels] = cp->subbands_to_free[levels]; >> + } >> + >> + memset(out[levels] + temp_length, 0, >> + (out_length[levels] - temp_length) * sizeof(**out)); >> + out_length[levels] = temp_length; >> + >> + return 0; >> +} >> + >> +static int discard_left_ext(int wavelet_length, int levels, int level, >> uint64_t sn) >> +{ >> + if (levels == level || sn == 0) >> + return 0; >> + return (pow2(levels - level) - 1) * (wavelet_length - 2) + >> mod_pow2(sn, pow2(levels)) / pow2(level); >> +} >> + >> +static int forward(AudioFWTDNContext *s, >> + const double *in, int in_length, >> + double **out, int *out_length, int ch, uint64_t sn) >> +{ >> + ChannelParams *cp = &s->cp[ch]; >> + int levels = s->levels; >> + int skip = sn ? s->wavelet_length - 1 : 1; >> + int leftext, ret; >> + >> + ret = reallocate_inputs(out, out_length, in_length, levels, ch, sn); >> + if (ret < 0) >> + return ret; >> + ret = reallocate_outputs(s, cp->filter_coefs, cp->filter_length, >> + in_length, levels, ch, sn); >> + if (ret < 0) >> + return ret; >> + >> + leftext = left_ext(s->wavelet_length, levels, sn); >> + >> + if (cp->temp_in_max_length < in_length + cp->max_left_ext + skip) { >> + av_freep(&cp->temp_in); >> + cp->temp_in_max_length = in_length + cp->max_left_ext + skip; >> + cp->temp_in = av_calloc(cp->temp_in_max_length, >> sizeof(*cp->temp_in)); >> + if (!cp->temp_in) { >> + cp->temp_in_max_length = 0; >> + return AVERROR(ENOMEM); >> + } >> + } >> + >> + memset(cp->temp_in, 0, cp->temp_in_max_length * >> sizeof(*cp->temp_in)); >> + cp->temp_in_length = in_length + leftext; >> + >> + if (leftext) >> + memcpy(cp->temp_in, cp->prev + s->prev_length - leftext, leftext >> * sizeof(*cp->temp_in)); >> + memcpy(cp->temp_in + leftext, in, in_length * sizeof(*in)); >> + >> + if (levels == 1) { >> + conv_down(cp->temp_in, cp->temp_in_length, out[1], out[0], >> out_length[1], >> + s->lp, s->hp, s->wavelet_length, skip, >> + cp->buffer, cp->buffer_length); >> + } else { >> + int discard = discard_left_ext(s->wavelet_length, levels, 1, >> sn); >> + int tempa_length_prev; >> + >> + if (cp->tempa_len_max < (in_length + cp->max_left_ext + >> s->wavelet_length - 1) / 2) { >> + av_freep(&cp->tempa); >> + av_freep(&cp->tempd); >> + cp->tempa_len_max = (in_length + cp->max_left_ext + >> s->wavelet_length - 1) / 2; >> + cp->tempa = av_calloc(cp->tempa_len_max, >> sizeof(*cp->tempa)); >> + cp->tempd = av_calloc(cp->tempa_len_max, >> sizeof(*cp->tempd)); >> + if (!cp->tempa || !cp->tempd) { >> + cp->tempa_len_max = 0; >> + return AVERROR(ENOMEM); >> + } >> + } >> + >> + memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa)); >> + memset(cp->tempd, 0, cp->tempa_len_max * sizeof(*cp->tempd)); >> + >> + cp->tempa_length = out_length[0] + discard; >> + conv_down(cp->temp_in, cp->temp_in_length, >> + cp->tempa, cp->tempd, cp->tempa_length, >> + s->lp, s->hp, s->wavelet_length, skip, >> + cp->buffer, cp->buffer_length); >> + memcpy(out[0], cp->tempd + discard, out_length[0] * >> sizeof(**out)); >> + tempa_length_prev = cp->tempa_length; >> + >> + for (int level = 1; level < levels - 1; level++) { >> + if (out_length[level] == 0) >> + return 0; >> + discard = discard_left_ext(s->wavelet_length, levels, level + >> 1, sn); >> + cp->tempa_length = out_length[level] + discard; >> + conv_down(cp->tempa, tempa_length_prev, >> + cp->tempa, cp->tempd, cp->tempa_length, >> + s->lp, s->hp, s->wavelet_length, skip, >> + cp->buffer, cp->buffer_length); >> + memcpy(out[level], cp->tempd + discard, out_length[level] * >> sizeof(**out)); >> + tempa_length_prev = cp->tempa_length; >> + } >> + >> + if (out_length[levels] == 0) >> + return 0; >> + conv_down(cp->tempa, cp->tempa_length, out[levels], out[levels - >> 1], out_length[levels], >> + s->lp, s->hp, s->wavelet_length, skip, >> + cp->buffer, cp->buffer_length); >> + } >> + >> + if (s->prev_length < in_length) { >> + memcpy(cp->prev, in + in_length - cp->max_left_ext, >> cp->max_left_ext * sizeof(*cp->prev)); >> + } else { >> + memmove(cp->prev, cp->prev + in_length, (s->prev_length - >> in_length) * sizeof(*cp->prev)); >> + memcpy(cp->prev + s->prev_length - in_length, in, in_length * >> sizeof(*cp->prev)); >> + } >> + >> + return 0; >> +} >> + >> +static void conv_up(double *low, double *high, int in_length, double >> *out, int out_length, >> + const double *lp, const double *hp, int >> filter_length, >> + double *buffer, double *buffer2, int buffer_length) >> +{ >> + int shift = 0, buff_idx = 0, in_idx = 0; >> + >> + memset(buffer, 0, buffer_length * sizeof(*buffer)); >> + memset(buffer2, 0, buffer_length * sizeof(*buffer2)); >> + >> + for (int i = 0; i < out_length; i++) { >> + double sum = 0.0; >> + >> + if ((i & 1) == 0) { >> + if (in_idx < in_length) { >> + buffer[buff_idx] = low[in_idx]; >> + buffer2[buff_idx] = high[in_idx++]; >> + } else { >> + buffer[buff_idx] = 0; >> + buffer2[buff_idx] = 0; >> + } >> + buff_idx++; >> + if (buff_idx >= buffer_length) >> + buff_idx = 0; >> + shift = 0; >> + } >> + >> + for (int j = 0; j < (filter_length - shift + 1) / 2; j++) { >> + const int idx = mod_pow2(-j + buff_idx - 1, buffer_length); >> + >> + sum += buffer[idx] * lp[j * 2 + shift] + buffer2[idx] * hp[j >> * 2 + shift]; >> + } >> + out[i] = sum; >> + shift = 1; >> + } >> +} >> + >> +static int append_left_ext(int wavelet_length, int levels, int level, >> uint64_t sn) >> +{ >> + if (levels == level) >> + return 0; >> + >> + return (pow2(levels - level) - 1) * (wavelet_length - 2) + >> + mod_pow2(sn, pow2(levels)) / pow2(level); >> +} >> + >> +static int inverse(AudioFWTDNContext *s, >> + double **in, int *in_length, >> + double *out, int out_length, int ch, uint64_t sn) >> +{ >> + ChannelParams *cp = &s->cp[ch]; >> + const int levels = s->levels; >> + int leftext = left_ext(s->wavelet_length, levels, sn); >> + int temp_skip = 0; >> + >> + if (sn == 0) >> + temp_skip = cp->min_left_ext; >> + >> + memset(out, 0, out_length * sizeof(*out)); >> + >> + if (cp->temp_in_max_length < out_length + cp->max_left_ext + >> s->wavelet_length - 1) { >> + av_freep(&cp->temp_in); >> + cp->temp_in_max_length = out_length + cp->max_left_ext + >> s->wavelet_length - 1; >> + cp->temp_in = av_calloc(cp->temp_in_max_length, >> sizeof(*cp->temp_in)); >> + if (!cp->temp_in) { >> + cp->temp_in_max_length = 0; >> + return AVERROR(ENOMEM); >> + } >> + } >> + >> + memset(cp->temp_in, 0, cp->temp_in_max_length * >> sizeof(*cp->temp_in)); >> + cp->temp_in_length = out_length + cp->max_left_ext; >> + >> + if (levels == 1) { >> + conv_up(in[1], in[0], in_length[1], cp->temp_in, >> cp->temp_in_length, >> + s->ilp, s->ihp, s->wavelet_length, >> + cp->buffer, cp->buffer2, cp->buffer_length); >> + memcpy(out + cp->max_left_ext - leftext, cp->temp_in + >> temp_skip, >> + FFMAX(0, out_length - (cp->max_left_ext - leftext)) * >> sizeof(*out)); >> + } else { >> + double *hp1, *hp2; >> + int add, add2; >> + >> + if (cp->tempa_len_max < (out_length + cp->max_left_ext + >> s->wavelet_length - 1) / 2) { >> + av_freep(&cp->tempa); >> + cp->tempa_len_max = (out_length + cp->max_left_ext + >> s->wavelet_length - 1) / 2; >> + cp->tempa = av_calloc(cp->tempa_len_max, >> sizeof(*cp->tempa)); >> + if (!cp->tempa) { >> + cp->tempa_len_max = 0; >> + return AVERROR(ENOMEM); >> + } >> + } >> + >> + memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa)); >> + >> + hp1 = levels & 1 ? cp->temp_in : cp->tempa; >> + hp2 = levels & 1 ? cp->tempa : cp->temp_in; >> + >> + add = append_left_ext(s->wavelet_length, levels, levels - 1, >> sn); >> + conv_up(in[levels], in[levels - 1], in_length[levels], hp1, >> in_length[levels - 2] + add, >> + s->ilp, s->ihp, s->wavelet_length, cp->buffer, >> cp->buffer2, cp->buffer_length); >> + >> + for (int level = levels - 1; level > 1; level--) { >> + add2 = append_left_ext(s->wavelet_length, levels, level - 1, >> sn); >> + add = append_left_ext(s->wavelet_length, levels, level, sn); >> + conv_up(hp1, in[level - 1] - add, in_length[level - 1] + >> add, >> + hp2, in_length[level - 2] + add2, >> + s->ilp, s->ihp, s->wavelet_length, >> + cp->buffer, cp->buffer2, cp->buffer_length); >> + FFSWAP(double *, hp1, hp2); >> + } >> + >> + add = append_left_ext(s->wavelet_length, levels, 1, sn); >> + conv_up(hp1, in[0] - add, in_length[0] + add, cp->temp_in, >> cp->temp_in_length, >> + s->ilp, s->ihp, s->wavelet_length, >> + cp->buffer, cp->buffer2, cp->buffer_length); >> + } >> + >> + memset(cp->temp_in, 0, temp_skip * sizeof(*cp->temp_in)); >> + if (s->overlap_length <= out_length) { >> + memcpy(out + cp->max_left_ext - leftext, cp->temp_in + >> temp_skip, >> + FFMAX(0, out_length - (cp->max_left_ext - leftext)) * >> sizeof(*out)); >> + for (int i = 0;i < FFMIN(s->overlap_length, out_length); i++) >> + out[i] += cp->overlap[i]; >> + >> + memcpy(cp->overlap, cp->temp_in + out_length - (cp->max_left_ext >> - leftext), >> + s->overlap_length * sizeof(*cp->overlap)); >> + } else { >> + for (int i = 0;i < s->overlap_length - (cp->max_left_ext - >> leftext); i++) >> + cp->overlap[i + cp->max_left_ext - leftext] += >> cp->temp_in[i]; >> + memcpy(out, cp->overlap, out_length * sizeof(*out)); >> + memmove(cp->overlap, cp->overlap + out_length, >> + (s->overlap_length - out_length) * >> sizeof(*cp->overlap)); >> + memcpy(cp->overlap + s->overlap_length - out_length, cp->temp_in >> + leftext, >> + out_length * sizeof(*cp->overlap)); >> + } >> + >> + return 0; >> +} >> + >> +static int next_pow2(int in) >> +{ >> + return 1 << (av_log2(in) + 1); >> +} >> + >> +static void denoise_level(double *out, const double *in, >> + const double *filter, >> + double percent, int length) >> +{ >> + const double x = percent * 0.01; >> + const double y = 1.0 - x; >> + >> + for (int i = 0; i < length; i++) >> + out[i] = x * filter[i] + in[i] * y; >> +} >> + >> +static double sqr(double in) >> +{ >> + return in * in; >> +} >> + >> +static double measure_mean(const double *in, int length) >> +{ >> + double sum = 0.0; >> + >> + for (int i = 0; i < length; i++) >> + sum += in[i]; >> + >> + return sum / length; >> +} >> + >> +static double measure_absmean(const double *in, int length) >> +{ >> + double sum = 0.0; >> + >> + for (int i = 0; i < length; i++) >> + sum += fabs(in[i]); >> + >> + return sum / length; >> +} >> + >> +static double measure_stddev(const double *in, int length, double mean) >> +{ >> + double sum = 0.; >> + >> + for (int i = 0; i < length; i++) { >> + sum += sqr(in[i] - mean); >> + } >> + >> + return sqrt(sum / length); >> +} >> + >> +static void noise_filter(const double stddev, const double *in, >> + double *out, double absmean, double softness, >> + double new_stddev, int length) >> +{ >> + for (int i = 0; i < length; i++) { >> + if (new_stddev <= stddev) >> + out[i] = 0.0; >> + else if (fabs(in[i]) <= absmean) >> + out[i] = 0.0; >> + else >> + out[i] = in[i] - FFSIGN(in[i]) * absmean / exp(3.0 * softness >> * (fabs(in[i]) - absmean) / absmean); >> + } >> +} >> + >> +typedef struct ThreadData { >> + AVFrame *in, *out; >> +} ThreadData; >> + >> +static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int >> nb_jobs) >> +{ >> + AudioFWTDNContext *s = ctx->priv; >> + ThreadData *td = arg; >> + AVFrame *in = td->in; >> + AVFrame *out = td->out; >> + ChannelParams *cp = &s->cp[ch]; >> + const double *src = (const double *)(in->extended_data[ch]); >> + double *dst = (double *)out->extended_data[ch]; >> + double *absmean = (double *)s->absmean->extended_data[ch]; >> + double *new_absmean = (double *)s->new_absmean->extended_data[ch]; >> + double *stddev = (double *)s->stddev->extended_data[ch]; >> + double *new_stddev = (double *)s->new_stddev->extended_data[ch]; >> + double *filter = (double *)s->filter->extended_data[ch]; >> + double is_noise = 0.0; >> + int ret; >> + >> + ret = forward(s, src, in->nb_samples, cp->output_coefs, >> cp->output_length, ch, s->sn); >> + if (ret < 0) >> + return ret; >> + >> + if (!s->got_profile && s->need_profile) { >> + for (int level = 0; level <= s->levels; level++) { >> + const int length = cp->output_length[level]; >> + const double scale = sqrt(2.0 * log(length)); >> + >> + stddev[level] = measure_stddev(cp->output_coefs[level], >> length, >> + measure_mean(cp->output_coefs[level], >> length)) * scale; >> + absmean[level] = measure_absmean(cp->output_coefs[level], >> length) * scale; >> + } >> + } else if (!s->got_profile && !s->need_profile && !s->adaptive) { >> + for (int level = 0; level <= s->levels; level++) { >> + const int length = cp->output_length[level]; >> + const double scale = sqrt(2.0 * log(length)); >> + >> + stddev[level] = 0.5 * s->sigma * scale; >> + absmean[level] = 0.5 * s->sigma * scale; >> + } >> + } >> + >> + for (int level = 0; level <= s->levels; level++) { >> + const int length = cp->output_length[level]; >> + double vad; >> + >> + new_stddev[level] = measure_stddev(cp->output_coefs[level], >> length, >> + measure_mean(cp->output_coefs[level], >> length)); >> + new_absmean[level] = measure_absmean(cp->output_coefs[level], >> length); >> + if (new_absmean[level] <= FLT_EPSILON) >> + vad = 1.0; >> + else >> + vad = new_stddev[level] / new_absmean[level]; >> + if (level < s->levels) >> + is_noise += sqr(vad - 1.232); >> + } >> + >> + is_noise *= in->sample_rate; >> + is_noise /= s->nb_samples; >> + for (int level = 0; level <= s->levels; level++) { >> + const int length = cp->output_length[level]; >> + const double scale = sqrt(2.0 * log(length)); >> + >> + if (is_noise < 0.05 && s->adaptive) { >> + stddev[level] = new_stddev[level] * scale; >> + absmean[level] = new_absmean[level] * scale; >> + } >> + >> + noise_filter(stddev[level], cp->output_coefs[level], filter, >> absmean[level], >> + s->softness, new_stddev[level], length); >> + denoise_level(cp->filter_coefs[level], cp->output_coefs[level], >> filter, s->percent, length); >> + } >> + >> + ret = inverse(s, cp->filter_coefs, cp->filter_length, dst, >> out->nb_samples, ch, s->sn); >> + if (ret < 0) >> + return ret; >> + >> + return 0; >> +} >> + >> +static int filter_frame(AVFilterLink *inlink, AVFrame *in) >> +{ >> + AVFilterContext *ctx = inlink->dst; >> + AudioFWTDNContext *s = ctx->priv; >> + AVFilterLink *outlink = ctx->outputs[0]; >> + ThreadData td; >> + AVFrame *out; >> + int eof = in == NULL; >> + >> + out = ff_get_audio_buffer(outlink, s->nb_samples); >> + if (!out) { >> + av_frame_free(&in); >> + return AVERROR(ENOMEM); >> + } >> + if (in) { >> + av_frame_copy_props(out, in); >> + s->eof_pts = in->pts + in->nb_samples; >> + } >> + if (eof) >> + out->pts = s->eof_pts - s->padd_samples; >> + >> + if (!in || in->nb_samples < s->nb_samples) { >> + AVFrame *new_in = ff_get_audio_buffer(outlink, s->nb_samples); >> + >> + if (!new_in) { >> + av_frame_free(&in); >> + av_frame_free(&out); >> + return AVERROR(ENOMEM); >> + } >> + if (in) >> + av_frame_copy_props(new_in, in); >> + >> + s->padd_samples -= s->nb_samples - (in ? in->nb_samples: 0); >> + if (in) >> + av_samples_copy(new_in->extended_data, in->extended_data, 0, >> 0, >> + in->nb_samples, in->channels, in->format); >> + av_frame_free(&in); >> + in = new_in; >> + } >> + >> + td.in = in; >> + td.out = out; >> + ctx->internal->execute(ctx, s->filter_channel, &td, NULL, >> inlink->channels); >> + if (s->need_profile) >> + s->got_profile = 1; >> + >> + s->sn += s->nb_samples; >> + >> + if (s->drop_samples >= in->nb_samples) { >> + s->drop_samples -= in->nb_samples; >> + s->delay += in->nb_samples; >> + av_frame_free(&in); >> + av_frame_free(&out); >> + FF_FILTER_FORWARD_STATUS(inlink, outlink); >> + FF_FILTER_FORWARD_WANTED(outlink, inlink); >> + return 0; >> + } else if (s->drop_samples > 0) { >> + for (int ch = 0; ch < out->channels; ch++) { >> + memmove(out->extended_data[ch], >> + out->extended_data[ch] + s->drop_samples * >> sizeof(double), >> + (in->nb_samples - s->drop_samples) * >> sizeof(double)); >> + } >> + >> + out->nb_samples = in->nb_samples - s->drop_samples; >> + out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, >> outlink->sample_rate}, outlink->time_base); >> + s->delay += s->drop_samples; >> + s->drop_samples = 0; >> + } else { >> + if (s->padd_samples < 0 && eof) { >> + out->nb_samples += s->padd_samples; >> + s->padd_samples = 0; >> + } >> + if (!eof) >> + out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, >> outlink->sample_rate}, outlink->time_base); >> + } >> + >> + av_frame_free(&in); >> + return ff_filter_frame(outlink, out); >> +} >> + >> +static int max_left_ext(int wavelet_length, int levels) >> +{ >> + return (pow2(levels) - 1) * (wavelet_length - 1); >> +} >> + >> +static int min_left_ext(int wavelet_length, int levels) >> +{ >> + return (pow2(levels) - 1) * (wavelet_length - 2); >> +} >> + >> +static int config_output(AVFilterLink *outlink) >> +{ >> + AVFilterContext *ctx = outlink->src; >> + AudioFWTDNContext *s = ctx->priv; >> + >> + switch (s->wavelet_type) { >> + case SYM2: >> + s->wavelet_length = 4; >> + s->lp = sym2_lp; >> + s->hp = sym2_hp; >> + s->ilp = sym2_ilp; >> + s->ihp = sym2_ihp; >> + break; >> + case SYM4: >> + s->wavelet_length = 8; >> + s->lp = sym4_lp; >> + s->hp = sym4_hp; >> + s->ilp = sym4_ilp; >> + s->ihp = sym4_ihp; >> + break; >> + case RBIOR68: >> + s->wavelet_length = 18; >> + s->lp = rbior68_lp; >> + s->hp = rbior68_hp; >> + s->ilp = rbior68_ilp; >> + s->ihp = rbior68_ihp; >> + break; >> + case DEB10: >> + s->wavelet_length = 20; >> + s->lp = deb10_lp; >> + s->hp = deb10_hp; >> + s->ilp = deb10_ilp; >> + s->ihp = deb10_ihp; >> + case SYM10: >> + s->wavelet_length = 20; >> + s->lp = sym10_lp; >> + s->hp = sym10_hp; >> + s->ilp = sym10_ilp; >> + s->ihp = sym10_ihp; >> + break; >> + case COIF5: >> + s->wavelet_length = 30; >> + s->lp = coif5_lp; >> + s->hp = coif5_hp; >> + s->ilp = coif5_ilp; >> + s->ihp = coif5_ihp; >> + break; >> + case BL3: >> + s->wavelet_length = 42; >> + s->lp = bl3_lp; >> + s->hp = bl3_hp; >> + s->ilp = bl3_ilp; >> + s->ihp = bl3_ihp; >> + break; >> + default: >> + av_assert0(0); >> + } >> + >> + s->levels = FFMIN(s->levels, lrint(log(s->nb_samples / >> (s->wavelet_length - 1.0)) / M_LN2)); >> + av_log(ctx, AV_LOG_VERBOSE, "levels: %d\n", s->levels); >> + s->filter_channel = filter_channel; >> + >> + s->stddev = ff_get_audio_buffer(outlink, MAX_LEVELS); >> + s->new_stddev = ff_get_audio_buffer(outlink, MAX_LEVELS); >> + s->filter = ff_get_audio_buffer(outlink, s->nb_samples); >> + s->absmean = ff_get_audio_buffer(outlink, MAX_LEVELS); >> + s->new_absmean = ff_get_audio_buffer(outlink, MAX_LEVELS); >> + if (!s->stddev || !s->absmean || !s->filter || >> + !s->new_stddev || !s->new_absmean) >> + return AVERROR(ENOMEM); >> + >> + s->channels = outlink->channels; >> + s->overlap_length = max_left_ext(s->wavelet_length, s->levels); >> + s->prev_length = s->overlap_length; >> + s->drop_samples = s->overlap_length; >> + s->padd_samples = s->overlap_length; >> + s->sn = 1; >> + >> + s->cp = av_calloc(s->channels, sizeof(*s->cp)); >> + if (!s->cp) >> + return AVERROR(ENOMEM); >> + >> + for (int ch = 0; ch < s->channels; ch++) { >> + ChannelParams *cp = &s->cp[ch]; >> + >> + cp->output_coefs = av_calloc(s->levels + 1, >> sizeof(*cp->output_coefs)); >> + cp->filter_coefs = av_calloc(s->levels + 1, >> sizeof(*cp->filter_coefs)); >> + cp->output_length = av_calloc(s->levels + 1, >> sizeof(*cp->output_length)); >> + cp->filter_length = av_calloc(s->levels + 1, >> sizeof(*cp->filter_length)); >> + cp->buffer_length = next_pow2(s->wavelet_length); >> + cp->buffer = av_calloc(cp->buffer_length, sizeof(*cp->buffer)); >> + cp->buffer2 = av_calloc(cp->buffer_length, >> sizeof(*cp->buffer2)); >> + cp->subbands_to_free = av_calloc(s->levels + 1, >> sizeof(*cp->subbands_to_free)); >> + cp->prev = av_calloc(s->prev_length, sizeof(*cp->prev)); >> + cp->overlap = av_calloc(s->overlap_length, >> sizeof(*cp->overlap)); >> + cp->max_left_ext = max_left_ext(s->wavelet_length, s->levels); >> + cp->min_left_ext = min_left_ext(s->wavelet_length, s->levels); >> + if (!cp->output_coefs || !cp->filter_coefs || !cp->output_length >> || >> + !cp->filter_length || !cp->subbands_to_free || !cp->prev || >> !cp->overlap || >> + !cp->buffer || !cp->buffer2) >> + return AVERROR(ENOMEM); >> + } >> + >> + return 0; >> +} >> + >> +static int activate(AVFilterContext *ctx) >> +{ >> + AVFilterLink *inlink = ctx->inputs[0]; >> + AVFilterLink *outlink = ctx->outputs[0]; >> + AudioFWTDNContext *s = ctx->priv; >> + AVFrame *in = NULL; >> + int ret, status; >> + int64_t pts; >> + >> + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); >> + >> + ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, >> &in); >> + if (ret < 0) >> + return ret; >> + if (ret > 0) >> + return filter_frame(inlink, in); >> + >> + if (ff_inlink_acknowledge_status(inlink, &status, &pts)) { >> + if (status == AVERROR_EOF) { >> + while (s->padd_samples != 0) { >> + ret = filter_frame(inlink, NULL); >> + if (ret < 0) >> + return ret; >> + } >> + ff_outlink_set_status(outlink, status, pts); >> + return ret; >> + } >> + } >> + FF_FILTER_FORWARD_WANTED(outlink, inlink); >> + >> + return FFERROR_NOT_READY; >> +} >> + >> +static av_cold void uninit(AVFilterContext *ctx) >> +{ >> + AudioFWTDNContext *s = ctx->priv; >> + >> + av_frame_free(&s->filter); >> + av_frame_free(&s->new_stddev); >> + av_frame_free(&s->stddev); >> + av_frame_free(&s->new_absmean); >> + av_frame_free(&s->absmean); >> + >> + for (int ch = 0; s->cp && ch < s->channels; ch++) { >> + ChannelParams *cp = &s->cp[ch]; >> + >> + av_freep(&cp->tempa); >> + av_freep(&cp->tempd); >> + av_freep(&cp->temp_in); >> + av_freep(&cp->buffer); >> + av_freep(&cp->buffer2); >> + av_freep(&cp->prev); >> + av_freep(&cp->overlap); >> + >> + av_freep(&cp->output_length); >> + av_freep(&cp->filter_length); >> + >> + if (cp->output_coefs) { >> + for (int level = 0; level <= s->levels; level++) >> + av_freep(&cp->output_coefs[level]); >> + } >> + >> + if (cp->subbands_to_free) { >> + for (int level = 0; level <= s->levels; level++) >> + av_freep(&cp->subbands_to_free[level]); >> + } >> + >> + av_freep(&cp->subbands_to_free); >> + av_freep(&cp->output_coefs); >> + av_freep(&cp->filter_coefs); >> + } >> + >> + av_freep(&s->cp); >> +} >> + >> +static int process_command(AVFilterContext *ctx, const char *cmd, const >> char *args, >> + char *res, int res_len, int flags) >> +{ >> + AudioFWTDNContext *s = ctx->priv; >> + int ret; >> + >> + ret = ff_filter_process_command(ctx, cmd, args, res, res_len, >> flags); >> + if (ret < 0) >> + return ret; >> + >> + if (!strcmp(cmd, "profile") && s->need_profile) >> + s->got_profile = 0; >> + >> + return 0; >> +} >> + >> +static const AVFilterPad inputs[] = { >> + { >> + .name = "default", >> + .type = AVMEDIA_TYPE_AUDIO, >> + }, >> + { NULL } >> +}; >> + >> +static const AVFilterPad outputs[] = { >> + { >> + .name = "default", >> + .type = AVMEDIA_TYPE_AUDIO, >> + .config_props = config_output, >> + }, >> + { NULL } >> +}; >> + >> +AVFilter ff_af_afwtdn = { >> + .name = "afwtdn", >> + .description = NULL_IF_CONFIG_SMALL("Denoise audio stream using >> Wavelets."), >> + .query_formats = query_formats, >> + .priv_size = sizeof(AudioFWTDNContext), >> + .priv_class = &afwtdn_class, >> + .activate = activate, >> + .uninit = uninit, >> + .inputs = inputs, >> + .outputs = outputs, >> + .process_command = process_command, >> + .flags = AVFILTER_FLAG_SLICE_THREADS, >> +}; >> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c >> index 1183e40267..de5884529c 100644 >> --- a/libavfilter/allfilters.c >> +++ b/libavfilter/allfilters.c >> @@ -43,6 +43,7 @@ extern AVFilter ff_af_afftdn; >> extern AVFilter ff_af_afftfilt; >> extern AVFilter ff_af_afir; >> extern AVFilter ff_af_aformat; >> +extern AVFilter ff_af_afwtdn; >> extern AVFilter ff_af_agate; >> extern AVFilter ff_af_aiir; >> extern AVFilter ff_af_aintegral; > > -- > Nicolas George >
Paul B Mahol (12020-06-12): > This is part of basic math, Then add the formula in a comment. But no, it is not just basic math: there was a program that did the computations. > you seems to lack basic math knowledge. Rude again.
On 6/12/20, Nicolas George <george@nsup.org> wrote: > Paul B Mahol (12020-06-12): >> This is part of basic math, > > Then add the formula in a comment. But no, it is not just basic math: > there was a program that did the computations. FFmpeg is not math encyclopedia. > >> you seems to lack basic math knowledge. > > Rude again. Your reviews are rude. > > -- > Nicolas George >
Paul B Mahol (12020-06-12):
> FFmpeg is not math encyclopedia.
You did not type these numbers, you took them from somewhere. Include it
in the source code, where it belongs.
On 12/06/2020 00:58, Nicolas George wrote: > Paul B Mahol (12020-06-12): >> FFmpeg is not math encyclopedia. > You did not type these numbers, you took them from somewhere. Include it > in the source code, where it belongs. I am not familiar with the entirety of this commit, but this seems to concern wavelet coefficients. I think linking a source for these coefficients that leads to further documentation instead of including this in the source code is a good idea. For anyone looking for such a source, http://wavelets.pybytes.com/wavelet/sym2/ offers a list of (rounded) coefficients (the specific link is for Symlets 2). Paul seems to have a source for either a more accurate list or the formulas to calculate these values. Linking this in the source might be really helpful for quick debugging or adapting the code later. I hope this comment was helpful in any way, -Valentin
On 7/23/20, Valentin Schweitzer <valentin.schweitzer98@gmail.com> wrote: > On 12/06/2020 00:58, Nicolas George wrote: >> Paul B Mahol (12020-06-12): >>> FFmpeg is not math encyclopedia. >> You did not type these numbers, you took them from somewhere. Include it >> in the source code, where it belongs. > > I am not familiar with the entirety of this commit, but this > seems to concern wavelet coefficients. I think linking a source > for these coefficients that leads to further documentation instead > of including this in the source code is a good idea. > > For anyone looking for such a source, > http://wavelets.pybytes.com/wavelet/sym2/ > offers a list of (rounded) coefficients (the specific link is for Symlets > 2). > Paul seems to have a source for either a more accurate list or the formulas > to > calculate these values. Linking this in the source might be really helpful > for > quick debugging or adapting the code later. > > I hope this comment was helpful in any way, LOL, very very funny. Site have javascript copy to copy coefficients, and yes for almost all wavelets I used that site as source. > > -Valentin > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
On 23/07/2020 14:55, Paul B Mahol wrote: > Site have javascript copy to copy coefficients, > and yes for almost all wavelets You're right, I missed that > I used that site as source. You could link it above the variable declaration, I think this might be useful for anyone looking for a source on that topic, or for anyone trying to contribute to this part of the code. Also, my prior e-mail may not have been plaintext, apologies for that. -Valentin
diff --git a/doc/filters.texi b/doc/filters.texi index c2960e33c7..d89ebc5122 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1314,6 +1314,66 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo aformat=sample_fmts=u8|s16:channel_layouts=stereo @end example +@section afwtdn +Reduce broadband noise from input samples using Wavelets. + +A description of the accepted options follows. + +@table @option +@item sigma +Set the noise sigma, allowed range is from 0 to 1. +Default value is 0. +This option controls strength of denoising applied to input samples. +Most useful way to set this option is via decibels, eg. -45dB. + +@item levels +Set the number of wavelet levels of decomposition. +Allowed range is from 1 to 12. +Default value is 10. +Setting this too low make denoising performance very poor. + +@item wavet +Set wavelet type for decomposition of input frame. +They are sorted by number of coefficients, from lowest to highest. +More coefficients means worse filtering speed, but overall better quality. +Available wavelets are: + +@table @samp +@item sym2 +@item sym4 +@item rbior68 +@item deb10 +@item sym10 +@item coif5 +@item bl3 +@end table + +@item percent +Set percent of full denoising. Allowed range is from 0 to 100 percent. +Default value is 85 percent or partial denoising. + +@item profile +If enabled, first input frame will be used as noise profile. +If first frame samples contain non-noise performance will be very poor. + +@item adaptive +If enabled, input frames are analyzed for presence of noise. +If noise is detected with high possibility then input frame profile will be +used for processing following frames, until new noise frame is detected. + +@item samples +Set size of single frame in number of samples. Allowed range is from 512 to +65536. Default frame size is 8192 samples. + +@item softness +Set softness applied inside thresholding function. Allowed range is from 0 to +10. Default softness is 1. +@end table + +@subsection Commands + +This filter supports subset of its options as @ref{commands}. + @section agate A gate is mainly used to reduce lower parts of a signal. This kind of signal diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 5123540653..191826a622 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -50,6 +50,7 @@ OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o +OBJS-$(CONFIG_AFWTDN_FILTER) += af_afwtdn.o OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o diff --git a/libavfilter/af_afwtdn.c b/libavfilter/af_afwtdn.c new file mode 100644 index 0000000000..d2793d4d92 --- /dev/null +++ b/libavfilter/af_afwtdn.c @@ -0,0 +1,1345 @@ +/* + * Copyright (c) 2020 Paul B Mahol + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <float.h> + +#include "libavutil/avassert.h" +#include "libavutil/avstring.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "filters.h" +#include "formats.h" + +enum WaveletTypes { + SYM2, + SYM4, + RBIOR68, + DEB10, + SYM10, + COIF5, + BL3, + NB_WAVELET_TYPES, +}; + +static const double bl3_lp[42] = { + 0.000146098, -0.000232304, -0.000285414, 0.000462093, 0.000559952, + -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426, + -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331, + 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147, + 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869, + 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477, + -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748, + -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304, + 0.000146098, 0.0, +}; + +static const double bl3_hp[42] = { + 0.0, 0.000146098, 0.000232304, -0.000285414, -0.000462093, 0.000559952, + 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426, + -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331, + -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147, + 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869, + -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477, + -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748, + 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304, + 0.000146098, +}; + +static const double bl3_ilp[42] = { + 0.0, 0.000146098, -0.000232304, -0.000285414, 0.000462093, 0.000559952, + -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426, + -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331, + 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147, + 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869, + 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477, + -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748, + -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304, + 0.000146098, +}; + +static const double bl3_ihp[42] = { + 0.000146098, 0.000232304, -0.000285414, -0.000462093, 0.000559952, + 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426, + -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331, + -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147, + 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869, + -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477, + -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748, + 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304, + 0.000146098, +}; + +static const double sym10_lp[20] = { + 0.0007701598091144901, 9.563267072289475e-05, + -0.008641299277022422, -0.0014653825813050513, + 0.0459272392310922, 0.011609893903711381, + -0.15949427888491757, -0.07088053578324385, + 0.47169066693843925, 0.7695100370211071, + 0.38382676106708546, -0.03553674047381755, + -0.0319900568824278, 0.04999497207737669, + 0.005764912033581909, -0.02035493981231129, + -0.0008043589320165449, 0.004593173585311828, + 5.7036083618494284e-05, -0.0004593294210046588, +}; + +static const double sym10_hp[20] = { + 0.0004593294210046588, 5.7036083618494284e-05, + -0.004593173585311828, -0.0008043589320165449, + 0.02035493981231129, 0.005764912033581909, + -0.04999497207737669, -0.0319900568824278, + 0.03553674047381755, 0.38382676106708546, + -0.7695100370211071, 0.47169066693843925, + 0.07088053578324385, -0.15949427888491757, + -0.011609893903711381, 0.0459272392310922, + 0.0014653825813050513, -0.008641299277022422, + -9.563267072289475e-05, 0.0007701598091144901, +}; + +static const double sym10_ilp[20] = { + -0.0004593294210046588, 5.7036083618494284e-05, + 0.004593173585311828, -0.0008043589320165449, + -0.02035493981231129, 0.005764912033581909, + 0.04999497207737669, -0.0319900568824278, + -0.03553674047381755, 0.38382676106708546, + 0.7695100370211071, 0.47169066693843925, + -0.07088053578324385, -0.15949427888491757, + 0.011609893903711381, 0.0459272392310922, + -0.0014653825813050513, -0.008641299277022422, + 9.563267072289475e-05, 0.0007701598091144901, +}; + +static const double sym10_ihp[20] = { + 0.0007701598091144901, -9.563267072289475e-05, + -0.008641299277022422, 0.0014653825813050513, + 0.0459272392310922, -0.011609893903711381, + -0.15949427888491757, 0.07088053578324385, + 0.47169066693843925, -0.7695100370211071, + 0.38382676106708546, 0.03553674047381755, + -0.0319900568824278, -0.04999497207737669, + 0.005764912033581909, 0.02035493981231129, + -0.0008043589320165449, -0.004593173585311828, + 5.7036083618494284e-05, 0.0004593294210046588, +}; + +static const double rbior68_lp[18] = { + 0.0, 0.0, 0.0, 0.0, + 0.014426282505624435, 0.014467504896790148, + -0.07872200106262882, -0.04036797903033992, + 0.41784910915027457, 0.7589077294536541, + 0.41784910915027457, -0.04036797903033992, + -0.07872200106262882, 0.014467504896790148, + 0.014426282505624435, 0.0, 0.0, 0.0, +}; + +static const double rbior68_hp[18] = { + -0.0019088317364812906, -0.0019142861290887667, + 0.016990639867602342, 0.01193456527972926, + -0.04973290349094079, -0.07726317316720414, + 0.09405920349573646, 0.4207962846098268, + -0.8259229974584023, 0.4207962846098268, + 0.09405920349573646, -0.07726317316720414, + -0.04973290349094079, 0.01193456527972926, + 0.016990639867602342, -0.0019142861290887667, + -0.0019088317364812906, 0.0, +}; + +static const double rbior68_ilp[18] = { + 0.0019088317364812906, -0.0019142861290887667, + -0.016990639867602342, 0.01193456527972926, + 0.04973290349094079, -0.07726317316720414, + -0.09405920349573646, 0.4207962846098268, + 0.8259229974584023, 0.4207962846098268, + -0.09405920349573646, -0.07726317316720414, + 0.04973290349094079, 0.01193456527972926, + -0.016990639867602342, -0.0019142861290887667, + 0.0019088317364812906, 0.0, +}; + +static const double rbior68_ihp[18] = { + 0.0, 0.0, 0.0, 0.0, + 0.014426282505624435, -0.014467504896790148, + -0.07872200106262882, 0.04036797903033992, + 0.41784910915027457, -0.7589077294536541, + 0.41784910915027457, 0.04036797903033992, + -0.07872200106262882, -0.014467504896790148, + 0.014426282505624435, 0.0, 0.0, 0.0, +}; + +static const double coif5_lp[30] = { + -9.517657273819165e-08, -1.6744288576823017e-07, + 2.0637618513646814e-06, 3.7346551751414047e-06, + -2.1315026809955787e-05, -4.134043227251251e-05, + 0.00014054114970203437, 0.00030225958181306315, + -0.0006381313430451114, -0.0016628637020130838, + 0.0024333732126576722, 0.006764185448053083, + -0.009164231162481846, -0.01976177894257264, + 0.03268357426711183, 0.0412892087501817, + -0.10557420870333893, -0.06203596396290357, + 0.4379916261718371, 0.7742896036529562, + 0.4215662066908515, -0.05204316317624377, + -0.09192001055969624, 0.02816802897093635, + 0.023408156785839195, -0.010131117519849788, + -0.004159358781386048, 0.0021782363581090178, + 0.00035858968789573785, -0.00021208083980379827, +}; + +static const double coif5_hp[30] = { + 0.00021208083980379827, 0.00035858968789573785, + -0.0021782363581090178, -0.004159358781386048, + 0.010131117519849788, 0.023408156785839195, + -0.02816802897093635, -0.09192001055969624, + 0.05204316317624377, 0.4215662066908515, + -0.7742896036529562, 0.4379916261718371, + 0.06203596396290357, -0.10557420870333893, + -0.0412892087501817, 0.03268357426711183, + 0.01976177894257264, -0.009164231162481846, + -0.006764185448053083, 0.0024333732126576722, + 0.0016628637020130838, -0.0006381313430451114, + -0.00030225958181306315, 0.00014054114970203437, + 4.134043227251251e-05, -2.1315026809955787e-05, + -3.7346551751414047e-06, 2.0637618513646814e-06, + 1.6744288576823017e-07, -9.517657273819165e-08, +}; + +static const double coif5_ilp[30] = { + -0.00021208083980379827, 0.00035858968789573785, + 0.0021782363581090178, -0.004159358781386048, + -0.010131117519849788, 0.023408156785839195, + 0.02816802897093635, -0.09192001055969624, + -0.05204316317624377, 0.4215662066908515, + 0.7742896036529562, 0.4379916261718371, + -0.06203596396290357, -0.10557420870333893, + 0.0412892087501817, 0.03268357426711183, + -0.01976177894257264, -0.009164231162481846, + 0.006764185448053083, 0.0024333732126576722, + -0.0016628637020130838, -0.0006381313430451114, + 0.00030225958181306315, 0.00014054114970203437, + -4.134043227251251e-05, -2.1315026809955787e-05, + 3.7346551751414047e-06, 2.0637618513646814e-06, + -1.6744288576823017e-07, -9.517657273819165e-08, +}; + +static const double coif5_ihp[30] = { + -9.517657273819165e-08, 1.6744288576823017e-07, + 2.0637618513646814e-06, -3.7346551751414047e-06, + -2.1315026809955787e-05, 4.134043227251251e-05, + 0.00014054114970203437, -0.00030225958181306315, + -0.0006381313430451114, 0.0016628637020130838, + 0.0024333732126576722, -0.006764185448053083, + -0.009164231162481846, 0.01976177894257264, + 0.03268357426711183, -0.0412892087501817, + -0.10557420870333893, 0.06203596396290357, + 0.4379916261718371, -0.7742896036529562, + 0.4215662066908515, 0.05204316317624377, + -0.09192001055969624, -0.02816802897093635, + 0.023408156785839195, 0.010131117519849788, + -0.004159358781386048, -0.0021782363581090178, + 0.00035858968789573785, 0.00021208083980379827, +}; + +static const double deb10_lp[20] = { + -1.326420300235487e-05, 9.358867000108985e-05, + -0.0001164668549943862, -0.0006858566950046825, + 0.00199240529499085, 0.0013953517469940798, + -0.010733175482979604, 0.0036065535669883944, + 0.03321267405893324, -0.02945753682194567, + -0.07139414716586077, 0.09305736460380659, + 0.12736934033574265, -0.19594627437659665, + -0.24984642432648865, 0.2811723436604265, + 0.6884590394525921, 0.5272011889309198, + 0.18817680007762133, 0.026670057900950818, +}; + +static const double deb10_hp[20] = { + -0.026670057900950818, 0.18817680007762133, + -0.5272011889309198, 0.6884590394525921, + -0.2811723436604265, -0.24984642432648865, + 0.19594627437659665, 0.12736934033574265, + -0.09305736460380659, -0.07139414716586077, + 0.02945753682194567, 0.03321267405893324, + -0.0036065535669883944, -0.010733175482979604, + -0.0013953517469940798, 0.00199240529499085, + 0.0006858566950046825, -0.0001164668549943862, + -9.358867000108985e-05, -1.326420300235487e-05, +}; + +static const double deb10_ilp[20] = { + 0.026670057900950818, 0.18817680007762133, + 0.5272011889309198, 0.6884590394525921, + 0.2811723436604265, -0.24984642432648865, + -0.19594627437659665, 0.12736934033574265, + 0.09305736460380659, -0.07139414716586077, + -0.02945753682194567, 0.03321267405893324, + 0.0036065535669883944, -0.010733175482979604, + 0.0013953517469940798, 0.00199240529499085, + -0.0006858566950046825, -0.0001164668549943862, + 9.358867000108985e-05, -1.326420300235487e-05, +}; + +static const double deb10_ihp[20] = { + -1.326420300235487e-05, -9.358867000108985e-05, + -0.0001164668549943862, 0.0006858566950046825, + 0.00199240529499085, -0.0013953517469940798, + -0.010733175482979604, -0.0036065535669883944, + 0.03321267405893324, 0.02945753682194567, + -0.07139414716586077, -0.09305736460380659, + 0.12736934033574265, 0.19594627437659665, + -0.24984642432648865, -0.2811723436604265, + 0.6884590394525921, -0.5272011889309198, + 0.18817680007762133, -0.026670057900950818, +}; + +static const double sym4_lp[8] = { + -0.07576571478927333, + -0.02963552764599851, + 0.49761866763201545, + 0.8037387518059161, + 0.29785779560527736, + -0.09921954357684722, + -0.012603967262037833, + 0.0322231006040427, +}; + +static const double sym4_hp[8] = { + -0.0322231006040427, + -0.012603967262037833, + 0.09921954357684722, + 0.29785779560527736, + -0.8037387518059161, + 0.49761866763201545, + 0.02963552764599851, + -0.07576571478927333, +}; + +static const double sym4_ilp[8] = { + 0.0322231006040427, + -0.012603967262037833, + -0.09921954357684722, + 0.29785779560527736, + 0.8037387518059161, + 0.49761866763201545, + -0.02963552764599851, + -0.07576571478927333, +}; + +static const double sym4_ihp[8] = { + -0.07576571478927333, + 0.02963552764599851, + 0.49761866763201545, + -0.8037387518059161, + 0.29785779560527736, + 0.09921954357684722, + -0.012603967262037833, + -0.0322231006040427, +}; + +static const double sym2_lp[4] = { + -0.12940952255092145, 0.22414386804185735, + 0.836516303737469, 0.48296291314469025, +}; + +static const double sym2_hp[4] = { + -0.48296291314469025, 0.836516303737469, + -0.22414386804185735, -0.12940952255092145, +}; + +static const double sym2_ilp[4] = { + 0.48296291314469025, 0.836516303737469, + 0.22414386804185735, -0.12940952255092145, +}; + +static const double sym2_ihp[4] = { + -0.12940952255092145, -0.22414386804185735, + 0.836516303737469, -0.48296291314469025, +}; + +#define MAX_LEVELS 13 + +typedef struct ChannelParams { + int *output_length; + int *filter_length; + double **output_coefs; + double **subbands_to_free; + double **filter_coefs; + + int tempa_length; + int tempa_len_max; + int temp_in_length; + int temp_in_max_length; + int buffer_length; + int min_left_ext; + int max_left_ext; + + double *tempa; + double *tempd; + double *temp_in; + double *buffer; + double *buffer2; + double *prev; + double *overlap; +} ChannelParams; + +typedef struct AudioFWTDNContext { + const AVClass *class; + + double sigma; + double percent; + double softness; + + uint64_t sn; + int64_t eof_pts; + + int wavelet_type; + int channels; + int nb_samples; + int levels; + int wavelet_length; + int need_profile; + int got_profile; + int adaptive; + + int delay; + int drop_samples; + int padd_samples; + int overlap_length; + int prev_length; + ChannelParams *cp; + + const double *lp, *hp; + const double *ilp, *ihp; + + AVFrame *stddev, *absmean, *filter; + AVFrame *new_stddev, *new_absmean; + + int (*filter_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs); +} AudioFWTDNContext; + +#define OFFSET(x) offsetof(AudioFWTDNContext, x) +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM +#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM + +static const AVOption afwtdn_options[] = { + { "sigma", "set noise sigma", OFFSET(sigma), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, AFR }, + { "levels", "set number of wavelet levels", OFFSET(levels), AV_OPT_TYPE_INT, {.i64=10}, 1, MAX_LEVELS-1, AF }, + { "wavet", "set wavelet type", OFFSET(wavelet_type), AV_OPT_TYPE_INT, {.i64=SYM10}, 0, NB_WAVELET_TYPES - 1, AF, "wavet" }, + { "sym2", "sym2", 0, AV_OPT_TYPE_CONST, {.i64=SYM2}, 0, 0, AF, "wavet" }, + { "sym4", "sym4", 0, AV_OPT_TYPE_CONST, {.i64=SYM4}, 0, 0, AF, "wavet" }, + { "rbior68", "rbior68", 0, AV_OPT_TYPE_CONST, {.i64=RBIOR68}, 0, 0, AF, "wavet" }, + { "deb10", "deb10", 0, AV_OPT_TYPE_CONST, {.i64=DEB10}, 0, 0, AF, "wavet" }, + { "sym10", "sym10", 0, AV_OPT_TYPE_CONST, {.i64=SYM10}, 0, 0, AF, "wavet" }, + { "coif5", "coif5", 0, AV_OPT_TYPE_CONST, {.i64=COIF5}, 0, 0, AF, "wavet" }, + { "bl3", "bl3", 0, AV_OPT_TYPE_CONST, {.i64=BL3}, 0, 0, AF, "wavet" }, + { "percent", "set percent of full denoising", OFFSET(percent),AV_OPT_TYPE_DOUBLE, {.dbl=85}, 0, 100, AFR }, + { "profile", "profile noise", OFFSET(need_profile), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AFR }, + { "adaptive", "adaptive profiling of noise", OFFSET(adaptive), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AFR }, + { "samples", "set frame size in number of samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=8192}, 512, 65536, AF }, + { "softness", "set thresholding softness", OFFSET(softness), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 10, AFR }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(afwtdn); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts = NULL; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + layouts = ff_all_channel_counts(); + if (!layouts) + return AVERROR(ENOMEM); + + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + return ff_set_common_samplerates(ctx, formats); +} + +#define pow2(x) (1U << (x)) +#define mod_pow2(x, power_of_two) ((x) & ((power_of_two) - 1)) + +static void conv_down(double *in, int in_length, double *low, double *high, + int out_length, const double *lp, const double *hp, + int wavelet_length, int skip, + double *buffer, int buffer_length) +{ + double thigh = 0.0, tlow = 0.0; + int buff_idx = 1 + skip; + + memcpy(buffer, in, buff_idx * sizeof(*buffer)); + memset(buffer + buff_idx, 0, (buffer_length - buff_idx) * sizeof(*buffer)); + + for (int i = 0; i < out_length - 1; i++) { + double thigh = 0.0, tlow = 0.0; + + for (int j = 0; j < wavelet_length; j++) { + const int idx = mod_pow2(-j + buff_idx - 1, buffer_length); + const double btemp = buffer[idx]; + + thigh += btemp * hp[j]; + tlow += btemp * lp[j]; + } + + high[i] = thigh; + low[i] = tlow; + buffer[buff_idx++] = in[2 * i + 1 + skip]; + buffer[buff_idx++] = in[2 * i + 2 + skip]; + buff_idx = mod_pow2(buff_idx, buffer_length); + } + + for (int i = 0; i < wavelet_length; i++) { + const int idx = mod_pow2(-i + buff_idx - 1, buffer_length); + const double btemp = buffer[idx]; + + thigh += btemp * hp[i]; + tlow += btemp * lp[i]; + } + + high[out_length - 1] = thigh; + low[out_length - 1] = tlow; +} + +static int left_ext(int wavelet_length, int levels, uint64_t sn) +{ + if (!sn) + return 0; + return (pow2(levels) - 1) * (wavelet_length - 2) + mod_pow2(sn, pow2(levels)); +} + +static int nb_coefs(int length, int level, uint64_t sn) +{ + const int pow2_level = pow2(level); + + return (sn + length) / pow2_level - sn / pow2_level; +} + +static int reallocate_inputs(double **out, int *out_length, + int in_length, int levels, int ch, uint64_t sn) +{ + const int temp_length = nb_coefs(in_length, levels, sn); + + for (int level = 0; level < levels; level++) { + const int temp_length = nb_coefs(in_length, level + 1, sn); + + if (temp_length > out_length[level]) { + av_freep(&out[level]); + out_length[level] = 0; + + out[level] = av_calloc(temp_length + 1, sizeof(**out)); + if (!out[level]) + return AVERROR(ENOMEM); + out_length[level] = temp_length + 1; + } + + memset(out[level] + temp_length, 0, + (out_length[level] - temp_length) * sizeof(**out)); + out_length[level] = temp_length; + } + + if (temp_length > out_length[levels]) { + av_freep(&out[levels]); + out_length[levels] = 0; + + out[levels] = av_calloc(temp_length + 1, sizeof(**out)); + if (!out[levels]) + return AVERROR(ENOMEM); + out_length[levels] = temp_length + 1; + } + + memset(out[levels] + temp_length, 0, + (out_length[levels] - temp_length) * sizeof(**out)); + out_length[levels] = temp_length; + + return 0; +} + +static int max_left_zeros_inverse(int levels, int level, int wavelet_length) +{ + return (pow2(levels - level) - 1) * (wavelet_length - 1); +} + +static int reallocate_outputs(AudioFWTDNContext *s, + double **out, int *out_length, + int in_length, int levels, int ch, uint64_t sn) +{ + ChannelParams *cp = &s->cp[ch]; + int temp_length = 0; + int add = 0; + + for (int level = 0; level < levels; level++) { + temp_length = nb_coefs(in_length, level + 1, sn); + if (temp_length > out_length[level]) { + av_freep(&cp->subbands_to_free[level]); + out_length[level] = 0; + + add = max_left_zeros_inverse(levels, level + 1, s->wavelet_length); + cp->subbands_to_free[level] = av_calloc(add + temp_length + 1, sizeof(**out)); + if (!cp->subbands_to_free[level]) + return AVERROR(ENOMEM); + out_length[level] = add + temp_length + 1; + out[level] = cp->subbands_to_free[level] + add; + } + + memset(out[level] + temp_length, 0, + FFMAX(out_length[level] - temp_length - add, 0) * sizeof(**out)); + out_length[level] = temp_length; + } + + temp_length = nb_coefs(in_length, levels, sn); + if (temp_length > out_length[levels]) { + av_freep(&cp->subbands_to_free[levels]); + out_length[levels] = 0; + + cp->subbands_to_free[levels] = av_calloc(temp_length + 1, sizeof(**out)); + if (!cp->subbands_to_free[levels]) + return AVERROR(ENOMEM); + out_length[levels] = temp_length + 1; + out[levels] = cp->subbands_to_free[levels]; + } + + memset(out[levels] + temp_length, 0, + (out_length[levels] - temp_length) * sizeof(**out)); + out_length[levels] = temp_length; + + return 0; +} + +static int discard_left_ext(int wavelet_length, int levels, int level, uint64_t sn) +{ + if (levels == level || sn == 0) + return 0; + return (pow2(levels - level) - 1) * (wavelet_length - 2) + mod_pow2(sn, pow2(levels)) / pow2(level); +} + +static int forward(AudioFWTDNContext *s, + const double *in, int in_length, + double **out, int *out_length, int ch, uint64_t sn) +{ + ChannelParams *cp = &s->cp[ch]; + int levels = s->levels; + int skip = sn ? s->wavelet_length - 1 : 1; + int leftext, ret; + + ret = reallocate_inputs(out, out_length, in_length, levels, ch, sn); + if (ret < 0) + return ret; + ret = reallocate_outputs(s, cp->filter_coefs, cp->filter_length, + in_length, levels, ch, sn); + if (ret < 0) + return ret; + + leftext = left_ext(s->wavelet_length, levels, sn); + + if (cp->temp_in_max_length < in_length + cp->max_left_ext + skip) { + av_freep(&cp->temp_in); + cp->temp_in_max_length = in_length + cp->max_left_ext + skip; + cp->temp_in = av_calloc(cp->temp_in_max_length, sizeof(*cp->temp_in)); + if (!cp->temp_in) { + cp->temp_in_max_length = 0; + return AVERROR(ENOMEM); + } + } + + memset(cp->temp_in, 0, cp->temp_in_max_length * sizeof(*cp->temp_in)); + cp->temp_in_length = in_length + leftext; + + if (leftext) + memcpy(cp->temp_in, cp->prev + s->prev_length - leftext, leftext * sizeof(*cp->temp_in)); + memcpy(cp->temp_in + leftext, in, in_length * sizeof(*in)); + + if (levels == 1) { + conv_down(cp->temp_in, cp->temp_in_length, out[1], out[0], out_length[1], + s->lp, s->hp, s->wavelet_length, skip, + cp->buffer, cp->buffer_length); + } else { + int discard = discard_left_ext(s->wavelet_length, levels, 1, sn); + int tempa_length_prev; + + if (cp->tempa_len_max < (in_length + cp->max_left_ext + s->wavelet_length - 1) / 2) { + av_freep(&cp->tempa); + av_freep(&cp->tempd); + cp->tempa_len_max = (in_length + cp->max_left_ext + s->wavelet_length - 1) / 2; + cp->tempa = av_calloc(cp->tempa_len_max, sizeof(*cp->tempa)); + cp->tempd = av_calloc(cp->tempa_len_max, sizeof(*cp->tempd)); + if (!cp->tempa || !cp->tempd) { + cp->tempa_len_max = 0; + return AVERROR(ENOMEM); + } + } + + memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa)); + memset(cp->tempd, 0, cp->tempa_len_max * sizeof(*cp->tempd)); + + cp->tempa_length = out_length[0] + discard; + conv_down(cp->temp_in, cp->temp_in_length, + cp->tempa, cp->tempd, cp->tempa_length, + s->lp, s->hp, s->wavelet_length, skip, + cp->buffer, cp->buffer_length); + memcpy(out[0], cp->tempd + discard, out_length[0] * sizeof(**out)); + tempa_length_prev = cp->tempa_length; + + for (int level = 1; level < levels - 1; level++) { + if (out_length[level] == 0) + return 0; + discard = discard_left_ext(s->wavelet_length, levels, level + 1, sn); + cp->tempa_length = out_length[level] + discard; + conv_down(cp->tempa, tempa_length_prev, + cp->tempa, cp->tempd, cp->tempa_length, + s->lp, s->hp, s->wavelet_length, skip, + cp->buffer, cp->buffer_length); + memcpy(out[level], cp->tempd + discard, out_length[level] * sizeof(**out)); + tempa_length_prev = cp->tempa_length; + } + + if (out_length[levels] == 0) + return 0; + conv_down(cp->tempa, cp->tempa_length, out[levels], out[levels - 1], out_length[levels], + s->lp, s->hp, s->wavelet_length, skip, + cp->buffer, cp->buffer_length); + } + + if (s->prev_length < in_length) { + memcpy(cp->prev, in + in_length - cp->max_left_ext, cp->max_left_ext * sizeof(*cp->prev)); + } else { + memmove(cp->prev, cp->prev + in_length, (s->prev_length - in_length) * sizeof(*cp->prev)); + memcpy(cp->prev + s->prev_length - in_length, in, in_length * sizeof(*cp->prev)); + } + + return 0; +} + +static void conv_up(double *low, double *high, int in_length, double *out, int out_length, + const double *lp, const double *hp, int filter_length, + double *buffer, double *buffer2, int buffer_length) +{ + int shift = 0, buff_idx = 0, in_idx = 0; + + memset(buffer, 0, buffer_length * sizeof(*buffer)); + memset(buffer2, 0, buffer_length * sizeof(*buffer2)); + + for (int i = 0; i < out_length; i++) { + double sum = 0.0; + + if ((i & 1) == 0) { + if (in_idx < in_length) { + buffer[buff_idx] = low[in_idx]; + buffer2[buff_idx] = high[in_idx++]; + } else { + buffer[buff_idx] = 0; + buffer2[buff_idx] = 0; + } + buff_idx++; + if (buff_idx >= buffer_length) + buff_idx = 0; + shift = 0; + } + + for (int j = 0; j < (filter_length - shift + 1) / 2; j++) { + const int idx = mod_pow2(-j + buff_idx - 1, buffer_length); + + sum += buffer[idx] * lp[j * 2 + shift] + buffer2[idx] * hp[j * 2 + shift]; + } + out[i] = sum; + shift = 1; + } +} + +static int append_left_ext(int wavelet_length, int levels, int level, uint64_t sn) +{ + if (levels == level) + return 0; + + return (pow2(levels - level) - 1) * (wavelet_length - 2) + + mod_pow2(sn, pow2(levels)) / pow2(level); +} + +static int inverse(AudioFWTDNContext *s, + double **in, int *in_length, + double *out, int out_length, int ch, uint64_t sn) +{ + ChannelParams *cp = &s->cp[ch]; + const int levels = s->levels; + int leftext = left_ext(s->wavelet_length, levels, sn); + int temp_skip = 0; + + if (sn == 0) + temp_skip = cp->min_left_ext; + + memset(out, 0, out_length * sizeof(*out)); + + if (cp->temp_in_max_length < out_length + cp->max_left_ext + s->wavelet_length - 1) { + av_freep(&cp->temp_in); + cp->temp_in_max_length = out_length + cp->max_left_ext + s->wavelet_length - 1; + cp->temp_in = av_calloc(cp->temp_in_max_length, sizeof(*cp->temp_in)); + if (!cp->temp_in) { + cp->temp_in_max_length = 0; + return AVERROR(ENOMEM); + } + } + + memset(cp->temp_in, 0, cp->temp_in_max_length * sizeof(*cp->temp_in)); + cp->temp_in_length = out_length + cp->max_left_ext; + + if (levels == 1) { + conv_up(in[1], in[0], in_length[1], cp->temp_in, cp->temp_in_length, + s->ilp, s->ihp, s->wavelet_length, + cp->buffer, cp->buffer2, cp->buffer_length); + memcpy(out + cp->max_left_ext - leftext, cp->temp_in + temp_skip, + FFMAX(0, out_length - (cp->max_left_ext - leftext)) * sizeof(*out)); + } else { + double *hp1, *hp2; + int add, add2; + + if (cp->tempa_len_max < (out_length + cp->max_left_ext + s->wavelet_length - 1) / 2) { + av_freep(&cp->tempa); + cp->tempa_len_max = (out_length + cp->max_left_ext + s->wavelet_length - 1) / 2; + cp->tempa = av_calloc(cp->tempa_len_max, sizeof(*cp->tempa)); + if (!cp->tempa) { + cp->tempa_len_max = 0; + return AVERROR(ENOMEM); + } + } + + memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa)); + + hp1 = levels & 1 ? cp->temp_in : cp->tempa; + hp2 = levels & 1 ? cp->tempa : cp->temp_in; + + add = append_left_ext(s->wavelet_length, levels, levels - 1, sn); + conv_up(in[levels], in[levels - 1], in_length[levels], hp1, in_length[levels - 2] + add, + s->ilp, s->ihp, s->wavelet_length, cp->buffer, cp->buffer2, cp->buffer_length); + + for (int level = levels - 1; level > 1; level--) { + add2 = append_left_ext(s->wavelet_length, levels, level - 1, sn); + add = append_left_ext(s->wavelet_length, levels, level, sn); + conv_up(hp1, in[level - 1] - add, in_length[level - 1] + add, + hp2, in_length[level - 2] + add2, + s->ilp, s->ihp, s->wavelet_length, + cp->buffer, cp->buffer2, cp->buffer_length); + FFSWAP(double *, hp1, hp2); + } + + add = append_left_ext(s->wavelet_length, levels, 1, sn); + conv_up(hp1, in[0] - add, in_length[0] + add, cp->temp_in, cp->temp_in_length, + s->ilp, s->ihp, s->wavelet_length, + cp->buffer, cp->buffer2, cp->buffer_length); + } + + memset(cp->temp_in, 0, temp_skip * sizeof(*cp->temp_in)); + if (s->overlap_length <= out_length) { + memcpy(out + cp->max_left_ext - leftext, cp->temp_in + temp_skip, + FFMAX(0, out_length - (cp->max_left_ext - leftext)) * sizeof(*out)); + for (int i = 0;i < FFMIN(s->overlap_length, out_length); i++) + out[i] += cp->overlap[i]; + + memcpy(cp->overlap, cp->temp_in + out_length - (cp->max_left_ext - leftext), + s->overlap_length * sizeof(*cp->overlap)); + } else { + for (int i = 0;i < s->overlap_length - (cp->max_left_ext - leftext); i++) + cp->overlap[i + cp->max_left_ext - leftext] += cp->temp_in[i]; + memcpy(out, cp->overlap, out_length * sizeof(*out)); + memmove(cp->overlap, cp->overlap + out_length, + (s->overlap_length - out_length) * sizeof(*cp->overlap)); + memcpy(cp->overlap + s->overlap_length - out_length, cp->temp_in + leftext, + out_length * sizeof(*cp->overlap)); + } + + return 0; +} + +static int next_pow2(int in) +{ + return 1 << (av_log2(in) + 1); +} + +static void denoise_level(double *out, const double *in, + const double *filter, + double percent, int length) +{ + const double x = percent * 0.01; + const double y = 1.0 - x; + + for (int i = 0; i < length; i++) + out[i] = x * filter[i] + in[i] * y; +} + +static double sqr(double in) +{ + return in * in; +} + +static double measure_mean(const double *in, int length) +{ + double sum = 0.0; + + for (int i = 0; i < length; i++) + sum += in[i]; + + return sum / length; +} + +static double measure_absmean(const double *in, int length) +{ + double sum = 0.0; + + for (int i = 0; i < length; i++) + sum += fabs(in[i]); + + return sum / length; +} + +static double measure_stddev(const double *in, int length, double mean) +{ + double sum = 0.; + + for (int i = 0; i < length; i++) { + sum += sqr(in[i] - mean); + } + + return sqrt(sum / length); +} + +static void noise_filter(const double stddev, const double *in, + double *out, double absmean, double softness, + double new_stddev, int length) +{ + for (int i = 0; i < length; i++) { + if (new_stddev <= stddev) + out[i] = 0.0; + else if (fabs(in[i]) <= absmean) + out[i] = 0.0; + else + out[i] = in[i] - FFSIGN(in[i]) * absmean / exp(3.0 * softness * (fabs(in[i]) - absmean) / absmean); + } +} + +typedef struct ThreadData { + AVFrame *in, *out; +} ThreadData; + +static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) +{ + AudioFWTDNContext *s = ctx->priv; + ThreadData *td = arg; + AVFrame *in = td->in; + AVFrame *out = td->out; + ChannelParams *cp = &s->cp[ch]; + const double *src = (const double *)(in->extended_data[ch]); + double *dst = (double *)out->extended_data[ch]; + double *absmean = (double *)s->absmean->extended_data[ch]; + double *new_absmean = (double *)s->new_absmean->extended_data[ch]; + double *stddev = (double *)s->stddev->extended_data[ch]; + double *new_stddev = (double *)s->new_stddev->extended_data[ch]; + double *filter = (double *)s->filter->extended_data[ch]; + double is_noise = 0.0; + int ret; + + ret = forward(s, src, in->nb_samples, cp->output_coefs, cp->output_length, ch, s->sn); + if (ret < 0) + return ret; + + if (!s->got_profile && s->need_profile) { + for (int level = 0; level <= s->levels; level++) { + const int length = cp->output_length[level]; + const double scale = sqrt(2.0 * log(length)); + + stddev[level] = measure_stddev(cp->output_coefs[level], length, + measure_mean(cp->output_coefs[level], length)) * scale; + absmean[level] = measure_absmean(cp->output_coefs[level], length) * scale; + } + } else if (!s->got_profile && !s->need_profile && !s->adaptive) { + for (int level = 0; level <= s->levels; level++) { + const int length = cp->output_length[level]; + const double scale = sqrt(2.0 * log(length)); + + stddev[level] = 0.5 * s->sigma * scale; + absmean[level] = 0.5 * s->sigma * scale; + } + } + + for (int level = 0; level <= s->levels; level++) { + const int length = cp->output_length[level]; + double vad; + + new_stddev[level] = measure_stddev(cp->output_coefs[level], length, + measure_mean(cp->output_coefs[level], length)); + new_absmean[level] = measure_absmean(cp->output_coefs[level], length); + if (new_absmean[level] <= FLT_EPSILON) + vad = 1.0; + else + vad = new_stddev[level] / new_absmean[level]; + if (level < s->levels) + is_noise += sqr(vad - 1.232); + } + + is_noise *= in->sample_rate; + is_noise /= s->nb_samples; + for (int level = 0; level <= s->levels; level++) { + const int length = cp->output_length[level]; + const double scale = sqrt(2.0 * log(length)); + + if (is_noise < 0.05 && s->adaptive) { + stddev[level] = new_stddev[level] * scale; + absmean[level] = new_absmean[level] * scale; + } + + noise_filter(stddev[level], cp->output_coefs[level], filter, absmean[level], + s->softness, new_stddev[level], length); + denoise_level(cp->filter_coefs[level], cp->output_coefs[level], filter, s->percent, length); + } + + ret = inverse(s, cp->filter_coefs, cp->filter_length, dst, out->nb_samples, ch, s->sn); + if (ret < 0) + return ret; + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AudioFWTDNContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + ThreadData td; + AVFrame *out; + int eof = in == NULL; + + out = ff_get_audio_buffer(outlink, s->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + if (in) { + av_frame_copy_props(out, in); + s->eof_pts = in->pts + in->nb_samples; + } + if (eof) + out->pts = s->eof_pts - s->padd_samples; + + if (!in || in->nb_samples < s->nb_samples) { + AVFrame *new_in = ff_get_audio_buffer(outlink, s->nb_samples); + + if (!new_in) { + av_frame_free(&in); + av_frame_free(&out); + return AVERROR(ENOMEM); + } + if (in) + av_frame_copy_props(new_in, in); + + s->padd_samples -= s->nb_samples - (in ? in->nb_samples: 0); + if (in) + av_samples_copy(new_in->extended_data, in->extended_data, 0, 0, + in->nb_samples, in->channels, in->format); + av_frame_free(&in); + in = new_in; + } + + td.in = in; + td.out = out; + ctx->internal->execute(ctx, s->filter_channel, &td, NULL, inlink->channels); + if (s->need_profile) + s->got_profile = 1; + + s->sn += s->nb_samples; + + if (s->drop_samples >= in->nb_samples) { + s->drop_samples -= in->nb_samples; + s->delay += in->nb_samples; + av_frame_free(&in); + av_frame_free(&out); + FF_FILTER_FORWARD_STATUS(inlink, outlink); + FF_FILTER_FORWARD_WANTED(outlink, inlink); + return 0; + } else if (s->drop_samples > 0) { + for (int ch = 0; ch < out->channels; ch++) { + memmove(out->extended_data[ch], + out->extended_data[ch] + s->drop_samples * sizeof(double), + (in->nb_samples - s->drop_samples) * sizeof(double)); + } + + out->nb_samples = in->nb_samples - s->drop_samples; + out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, outlink->sample_rate}, outlink->time_base); + s->delay += s->drop_samples; + s->drop_samples = 0; + } else { + if (s->padd_samples < 0 && eof) { + out->nb_samples += s->padd_samples; + s->padd_samples = 0; + } + if (!eof) + out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, outlink->sample_rate}, outlink->time_base); + } + + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static int max_left_ext(int wavelet_length, int levels) +{ + return (pow2(levels) - 1) * (wavelet_length - 1); +} + +static int min_left_ext(int wavelet_length, int levels) +{ + return (pow2(levels) - 1) * (wavelet_length - 2); +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioFWTDNContext *s = ctx->priv; + + switch (s->wavelet_type) { + case SYM2: + s->wavelet_length = 4; + s->lp = sym2_lp; + s->hp = sym2_hp; + s->ilp = sym2_ilp; + s->ihp = sym2_ihp; + break; + case SYM4: + s->wavelet_length = 8; + s->lp = sym4_lp; + s->hp = sym4_hp; + s->ilp = sym4_ilp; + s->ihp = sym4_ihp; + break; + case RBIOR68: + s->wavelet_length = 18; + s->lp = rbior68_lp; + s->hp = rbior68_hp; + s->ilp = rbior68_ilp; + s->ihp = rbior68_ihp; + break; + case DEB10: + s->wavelet_length = 20; + s->lp = deb10_lp; + s->hp = deb10_hp; + s->ilp = deb10_ilp; + s->ihp = deb10_ihp; + case SYM10: + s->wavelet_length = 20; + s->lp = sym10_lp; + s->hp = sym10_hp; + s->ilp = sym10_ilp; + s->ihp = sym10_ihp; + break; + case COIF5: + s->wavelet_length = 30; + s->lp = coif5_lp; + s->hp = coif5_hp; + s->ilp = coif5_ilp; + s->ihp = coif5_ihp; + break; + case BL3: + s->wavelet_length = 42; + s->lp = bl3_lp; + s->hp = bl3_hp; + s->ilp = bl3_ilp; + s->ihp = bl3_ihp; + break; + default: + av_assert0(0); + } + + s->levels = FFMIN(s->levels, lrint(log(s->nb_samples / (s->wavelet_length - 1.0)) / M_LN2)); + av_log(ctx, AV_LOG_VERBOSE, "levels: %d\n", s->levels); + s->filter_channel = filter_channel; + + s->stddev = ff_get_audio_buffer(outlink, MAX_LEVELS); + s->new_stddev = ff_get_audio_buffer(outlink, MAX_LEVELS); + s->filter = ff_get_audio_buffer(outlink, s->nb_samples); + s->absmean = ff_get_audio_buffer(outlink, MAX_LEVELS); + s->new_absmean = ff_get_audio_buffer(outlink, MAX_LEVELS); + if (!s->stddev || !s->absmean || !s->filter || + !s->new_stddev || !s->new_absmean) + return AVERROR(ENOMEM); + + s->channels = outlink->channels; + s->overlap_length = max_left_ext(s->wavelet_length, s->levels); + s->prev_length = s->overlap_length; + s->drop_samples = s->overlap_length; + s->padd_samples = s->overlap_length; + s->sn = 1; + + s->cp = av_calloc(s->channels, sizeof(*s->cp)); + if (!s->cp) + return AVERROR(ENOMEM); + + for (int ch = 0; ch < s->channels; ch++) { + ChannelParams *cp = &s->cp[ch]; + + cp->output_coefs = av_calloc(s->levels + 1, sizeof(*cp->output_coefs)); + cp->filter_coefs = av_calloc(s->levels + 1, sizeof(*cp->filter_coefs)); + cp->output_length = av_calloc(s->levels + 1, sizeof(*cp->output_length)); + cp->filter_length = av_calloc(s->levels + 1, sizeof(*cp->filter_length)); + cp->buffer_length = next_pow2(s->wavelet_length); + cp->buffer = av_calloc(cp->buffer_length, sizeof(*cp->buffer)); + cp->buffer2 = av_calloc(cp->buffer_length, sizeof(*cp->buffer2)); + cp->subbands_to_free = av_calloc(s->levels + 1, sizeof(*cp->subbands_to_free)); + cp->prev = av_calloc(s->prev_length, sizeof(*cp->prev)); + cp->overlap = av_calloc(s->overlap_length, sizeof(*cp->overlap)); + cp->max_left_ext = max_left_ext(s->wavelet_length, s->levels); + cp->min_left_ext = min_left_ext(s->wavelet_length, s->levels); + if (!cp->output_coefs || !cp->filter_coefs || !cp->output_length || + !cp->filter_length || !cp->subbands_to_free || !cp->prev || !cp->overlap || + !cp->buffer || !cp->buffer2) + return AVERROR(ENOMEM); + } + + return 0; +} + +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + AudioFWTDNContext *s = ctx->priv; + AVFrame *in = NULL; + int ret, status; + int64_t pts; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in); + if (ret < 0) + return ret; + if (ret > 0) + return filter_frame(inlink, in); + + if (ff_inlink_acknowledge_status(inlink, &status, &pts)) { + if (status == AVERROR_EOF) { + while (s->padd_samples != 0) { + ret = filter_frame(inlink, NULL); + if (ret < 0) + return ret; + } + ff_outlink_set_status(outlink, status, pts); + return ret; + } + } + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioFWTDNContext *s = ctx->priv; + + av_frame_free(&s->filter); + av_frame_free(&s->new_stddev); + av_frame_free(&s->stddev); + av_frame_free(&s->new_absmean); + av_frame_free(&s->absmean); + + for (int ch = 0; s->cp && ch < s->channels; ch++) { + ChannelParams *cp = &s->cp[ch]; + + av_freep(&cp->tempa); + av_freep(&cp->tempd); + av_freep(&cp->temp_in); + av_freep(&cp->buffer); + av_freep(&cp->buffer2); + av_freep(&cp->prev); + av_freep(&cp->overlap); + + av_freep(&cp->output_length); + av_freep(&cp->filter_length); + + if (cp->output_coefs) { + for (int level = 0; level <= s->levels; level++) + av_freep(&cp->output_coefs[level]); + } + + if (cp->subbands_to_free) { + for (int level = 0; level <= s->levels; level++) + av_freep(&cp->subbands_to_free[level]); + } + + av_freep(&cp->subbands_to_free); + av_freep(&cp->output_coefs); + av_freep(&cp->filter_coefs); + } + + av_freep(&s->cp); +} + +static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, + char *res, int res_len, int flags) +{ + AudioFWTDNContext *s = ctx->priv; + int ret; + + ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); + if (ret < 0) + return ret; + + if (!strcmp(cmd, "profile") && s->need_profile) + s->got_profile = 0; + + return 0; +} + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, + { NULL } +}; + +AVFilter ff_af_afwtdn = { + .name = "afwtdn", + .description = NULL_IF_CONFIG_SMALL("Denoise audio stream using Wavelets."), + .query_formats = query_formats, + .priv_size = sizeof(AudioFWTDNContext), + .priv_class = &afwtdn_class, + .activate = activate, + .uninit = uninit, + .inputs = inputs, + .outputs = outputs, + .process_command = process_command, + .flags = AVFILTER_FLAG_SLICE_THREADS, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 1183e40267..de5884529c 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -43,6 +43,7 @@ extern AVFilter ff_af_afftdn; extern AVFilter ff_af_afftfilt; extern AVFilter ff_af_afir; extern AVFilter ff_af_aformat; +extern AVFilter ff_af_afwtdn; extern AVFilter ff_af_agate; extern AVFilter ff_af_aiir; extern AVFilter ff_af_aintegral;
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- doc/filters.texi | 60 ++ libavfilter/Makefile | 1 + libavfilter/af_afwtdn.c | 1345 ++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 1407 insertions(+) create mode 100644 libavfilter/af_afwtdn.c