On 6/11/20, Nicolas George <geo...@nsup.org> wrote:
> Paul B Mahol (12020-06-11):
>> Signed-off-by: Paul B Mahol <one...@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
>
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