On 6/2/20, Nicolas George <geo...@nsup.org> wrote:
> Paul B Mahol (12020-06-02):
>> Signed-off-by: Paul B Mahol <one...@gmail.com>
>> ---
>> doc/filters.texi | 41 +++
>> libavfilter/Makefile | 1 +
>
>> libavfilter/af_afwtdn.c | 653 +++++++++++++++++++++++++++++++++++++++
>
> I still oppose strongly to the name of the filter.
Your opinion is not technical one.
>
>> libavfilter/allfilters.c | 1 +
>> 4 files changed, 696 insertions(+)
>> create mode 100644 libavfilter/af_afwtdn.c
>>
>> diff --git a/doc/filters.texi b/doc/filters.texi
>> index a0b4ab2228..15a8636398 100644
>> --- a/doc/filters.texi
>> +++ b/doc/filters.texi
>> @@ -1314,6 +1314,47 @@ 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 Fast Wavelet Transform.
>> +
>> +A description of the accepted options follows.
>> +
>> +@table @option
>> +@item sigma
>> +Set the noise sigma, allowed range is from 0 to 100.
>> +Default value is 0.
>> +This option controls strength of denoising applied to input samples.
>> +
>> +@item levels
>> +Set the number of wavelet levels of decomposition.
>> +Allowed range is from 1 to 19.
>> +Default value is 10.
>> +Setting this too low make denoising performance very poor.
>> +
>> +@item wavet
>> +Set wavelet type for decomposition of input frame.
>> +Available wavelets are:
>> +
>
>> +@table @samp
>> +@item deb4
>> +@item deb10
>> +@item coif5
>> +@item rbior68
>> +@end table
>
> Insufficient: users have no idea what it means.
They need only to experiment, we can not teach them whole wavelet
theory in 5 minutes.
>
>> +
>> +@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 set enabled, first input frame will be used as noise profile.
>> +If first frame samples contain non-noise performance will be very poor.
>> +
>
>> +@item frames
>> +Set size of single frame in number of samples. Allowed range is from 1024
>> to
>> +262144. Default frame size is 8192 samples.
>
> If it is the size of a single frame, why is it in plural?
frames stands for _frame_ _s_ize.
>
>> +@end table
>> +
>> @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..2d51c7bf2d
>> --- /dev/null
>> +++ b/libavfilter/af_afwtdn.c
>> @@ -0,0 +1,653 @@
>> +/*
>> + * 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/audio_fifo.h"
>> +#include "libavutil/avstring.h"
>> +#include "libavutil/opt.h"
>> +#include "avfilter.h"
>> +#include "audio.h"
>> +#include "filters.h"
>> +#include "formats.h"
>> +
>
>> +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 deb4_lp[8] = {
>> + -0.0105974018, 0.0328830117,
>> + 0.0308413818, -0.1870348117,
>> + -0.0279837694, 0.6308807679,
>> + 0.7148465706, 0.2303778133,
>> +};
>> +
>> +static const double deb4_hp[8] = {
>> + -0.2303778133, 0.7148465706,
>> + -0.6308807679, -0.0279837694,
>> + 0.1870348117, 0.0308413818,
>> + -0.0328830117, -0.0105974018,
>> +};
>> +
>> +static const double deb4_ilp[8] = {
>> + 0.23037781330885523, 0.7148465705525415,
>> + 0.6308807679295904, -0.02798376941698385,
>> + -0.18703481171888114, 0.030841381835986965,
>> + 0.032883011666982945, -0.010597401784997278,
>> +};
>> +
>> +static const double deb4_ihp[8] = {
>> + -0.010597401784997278, -0.032883011666982945,
>> + 0.030841381835986965, 0.18703481171888114,
>> + -0.02798376941698385, -0.6308807679295904,
>> + 0.7148465705525415, -0.23037781330885523,
>> +};
>
> A bunch of magical numbers. We need to know where they come from.
>
Get back to math class, those number are not magical at all.
Seriously there are other wavelet non-audio filters in libavfilter or
even libavcodec already, and they do not show from where numbers come.
>> +
>> +#define MAX_LEVELS 20
>> +
>> +typedef struct AudioFWTDNContext {
>> + const AVClass *class;
>> +
>> + double sigma;
>> + double percent;
>> +
>> + int wavelet_type;
>> + int thresholding_function;
>> + int nb_samples;
>> + int levels;
>> + int wavelet_length;
>> + int length[MAX_LEVELS];
>> + int out_length;
>> + int need_profile;
>> + int got_profile;
>> +
>> + const double *lp, *hp;
>> + const double *ilp, *ihp;
>> +
>> + AVFrame *temp0, *temp1;
>> + AVFrame *wtc, *power, *profile, *filter, *signal;
>> +} AudioFWTDNContext;
>> +
>> +#define OFFSET(x) offsetof(AudioFWTDNContext, x)
>> +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
>> +
>> +static const AVOption afwtdn_options[] = {
>> + { "sigma", "set noise sigma", OFFSET(sigma), AV_OPT_TYPE_DOUBLE,
>> {.dbl=0}, 0, 100, AF },
>> + { "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=0}, 0, 3, AF, "wavet" },
>> + { "deb4", "deb4", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "wavet"
>> },
>> + { "deb10", "deb10", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF,
>> "wavet" },
>> + { "coif5", "coif5", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF,
>> "wavet" },
>> + { "rbior68", "rbior68", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF,
>> "wavet" },
>> + { "percent", "set percent of full denoising",
>> OFFSET(percent),AV_OPT_TYPE_DOUBLE, {.dbl=85}, 0, 100, AF },
>> + { "profile", "profile noise", OFFSET(need_profile), AV_OPT_TYPE_BOOL,
>> {.i64=0}, 0, 1, AF },
>> + { "frames", "set frame size", OFFSET(nb_samples), AV_OPT_TYPE_INT,
>> {.i64=8192}, 1024, 262144, AF },
>> + { 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);
>> +}
>> +
>> +static int config_output(AVFilterLink *outlink)
>> +{
>> + AVFilterContext *ctx = outlink->src;
>> + AudioFWTDNContext *s = ctx->priv;
>> + int N, i;
>> +
>> + switch (s->wavelet_type) {
>> + case 0:
>> + s->wavelet_length = 8;
>> + s->lp = deb4_lp;
>> + s->hp = deb4_hp;
>> + s->ilp = deb4_ilp;
>> + s->ihp = deb4_ihp;
>> + break;
>> + case 1:
>> + s->wavelet_length = 20;
>> + s->lp = deb10_lp;
>> + s->hp = deb10_hp;
>> + s->ilp = deb10_ilp;
>> + s->ihp = deb10_ihp;
>> + break;
>> + case 2:
>> + s->wavelet_length = 30;
>> + s->lp = coif5_lp;
>> + s->hp = coif5_hp;
>> + s->ilp = coif5_ilp;
>> + s->ihp = coif5_ihp;
>> + break;
>> + case 3:
>> + s->wavelet_length = 18;
>> + s->lp = rbior68_lp;
>> + s->hp = rbior68_hp;
>> + s->ilp = rbior68_ilp;
>> + s->ihp = rbior68_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);
>> + N = s->nb_samples;
>> + i = s->levels;
>> +
>> + while (i > 0) {
>> + N = N + s->wavelet_length - 2;
>> + N = ceil(N / 2.0);
>> + s->length[i] = N;
>> + s->out_length += N;
>> + i--;
>> + }
>> +
>> + s->length[0] = s->length[1];
>> + s->out_length += s->length[0];
>> +
>> + s->temp0 = ff_get_audio_buffer(outlink, s->out_length);
>> + s->temp1 = ff_get_audio_buffer(outlink, s->out_length);
>> + s->wtc = ff_get_audio_buffer(outlink, s->out_length);
>> + s->power = ff_get_audio_buffer(outlink, s->out_length);
>> + s->filter = ff_get_audio_buffer(outlink, s->out_length);
>> + s->signal = ff_get_audio_buffer(outlink, s->out_length);
>> + s->profile = ff_get_audio_buffer(outlink, MAX_LEVELS);
>> + if (!s->temp0 || !s->temp1 || !s->wtc ||
>> + !s->power || !s->profile || !s->filter ||
>> + !s->signal)
>> + return AVERROR(ENOMEM);
>> +
>> + return 0;
>> +}
>> +
>> +static void dwt(double *src, int N,
>> + const double *lf, const double *hf, int flen,
>> + double *low, double *high, int lhlen)
>> +{
>> + for (int i = 0; i < lhlen; i++) {
>> + const int t = 2 * i + 1;
>> +
>> + low[i] = 0.0;
>> + high[i] = 0.0;
>> +
>> + for (int l = 0; l < flen; l++) {
>> + if (t - l >= 0 && t - l < N) {
>> + const int is = t - l;
>> + low[i] += lf[l] * src[is];
>> + high[i] += hf[l] * src[is];
>> + } else if (t - l < 0) {
>> + const int is = -t + l - 1;
>> + low[i] += lf[l] * src[is];
>> + high[i] += hf[l] * src[is];
>> + } else if (t - l >= N) {
>> + const int is = 2 * N - t + l - 1;
>> + low[i] += lf[l] * src[is];
>> + high[i] += hf[l] * src[is];
>> + }
>> + }
>> + }
>> +}
>> +
>> +static void idwt(double *low, int lhlen, double *high,
>> + const double *lf, const double *hf,
>> + int flen, double *dst)
>> +{
>> + int m = -2, n = -1;
>> +
>> + for (int v = 0; v < lhlen; v++) {
>> + m += 2;
>> + n += 2;
>> + dst[m] = 0.0;
>> + dst[n] = 0.0;
>> + for (int l = 0; l < flen / 2; l++) {
>> + const int t = 2 * l;
>> +
>> + if (v - l >= 0 && v - l < lhlen) {
>> + const int is = v - l;
>> +
>> + dst[m] += lf[t] * low[is] + hf[t] * high[is];
>> + dst[n] += lf[t + 1] * low[is] + hf[t + 1] * high[is];
>> + }
>> + }
>> + }
>> +}
>> +
>> +static void dwt_levels(AudioFWTDNContext *s, int levels, int inlength,
>> int out_length,
>> + const double *in, double *temp0, double *temp1,
>> double *wtc)
>> +{
>> + int N = out_length;
>> + int temp_len = s->nb_samples;
>> +
>> + for (int i = 0; i < inlength; i++)
>> + temp0[i] = in[i];
>> + for (int i = inlength; i < s->nb_samples; i++)
>> + temp0[i] = 0.;
>> +
>> + for (int level = 0; level < levels; level++) {
>> + const int level_length = s->length[levels - level];
>> +
>> + N -= level_length;
>> + dwt(temp0, temp_len, s->lp, s->hp, s->wavelet_length, temp1, wtc
>> + N, level_length);
>> + temp_len = s->length[levels - level];
>> +
>> + if (level == levels - 1) {
>> + for (int i = 0; i < level_length; i++)
>> + wtc[i] = temp1[i];
>> + } else {
>> + for (int i = 0; i < level_length; i++)
>> + temp0[i] = temp1[i];
>> + }
>> + }
>> +}
>> +
>> +static void idwt_levels(AudioFWTDNContext *s, int levels, double *out,
>> + double *temp, double *wtc)
>> +{
>> + const int app_len = s->length[0];
>> + const int lf = s->wavelet_length;
>> + int iter = app_len;
>> +
>> + for (int i = 0; i < app_len; i++)
>> + out[i] = wtc[i];
>> +
>> + for (int i = 0; i < levels; i++) {
>> + const int det_len = s->length[i + 1];
>> +
>> + idwt(out, det_len, wtc + iter, s->ilp, s->ihp, s->wavelet_length,
>> temp);
>> + for (int k = lf - 2; k < 2 * det_len; k++)
>> + out[k - lf + 2] = temp[k];
>> +
>> + iter += det_len;
>> + }
>> +}
>> +
>> +static void denoise_level(double *out, const double *in, double *filter,
>> double *signal, 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] = in[i] * (x * filter[i] * signal[i] + y);
>> +}
>> +
>> +static double sqr(double in)
>> +{
>> + return in * in;
>> +}
>> +
>> +static void power_average(const double *in, double *out, int length)
>> +{
>> + out[0] = sqr(in[0]);
>> + for (int i = 1; i < length; i++)
>> + out[i] = 0.1 * sqr(in[i]) + 0.9 * out[i - 1];
>> +}
>> +
>> +static void noise_filter(const double *in, double *out, double profile,
>> double ak, int length)
>> +{
>> + for (int i = 0; i < length; i++)
>> + out[i] = 1.0 - profile * ak / in[i];
>> +}
>> +
>> +static void signal_filter(const double *in, double *out, double profile,
>> double ak, int length)
>> +{
>> + for (int i = 0; i < length; i++)
>> + out[i] = in[i] >= profile * ak;
>> +}
>> +
>> +static void measure_level_profile(AVFilterContext *ctx, const double
>> *wtc, double *profile, int length)
>> +{
>> + double sum = 0.;
>> +
>> + for (int i = 0; i < length; i++)
>> + sum += sqr(wtc[i]);
>> +
>> + *profile = sum / length;
>> +}
>> +
>> +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;
>> + const double *src = (const double *)(in->extended_data[ch]);
>> + double *temp0 = (double *)s->temp0->extended_data[ch];
>> + double *temp1 = (double *)s->temp1->extended_data[ch];
>> + double *wtc = (double *)s->wtc->extended_data[ch];
>> + double *dst = (double *)out->extended_data[ch];
>> + double *profile = (double *)s->profile->extended_data[ch];
>> + double *power = (double *)s->power->extended_data[ch];
>> + double *filter = (double *)s->filter->extended_data[ch];
>> + double *signal = (double *)s->signal->extended_data[ch];
>> + int offset = 0;
>> +
>> + dwt_levels(s, s->levels, in->nb_samples, s->out_length, src, temp0,
>> temp1, wtc);
>> +
>> + if (!s->got_profile && s->need_profile) {
>> + for (int level = 0; level <= s->levels; level++) {
>> + measure_level_profile(ctx, wtc + offset, &profile[level],
>> s->length[level]);
>> + offset += s->length[level];
>> + }
>> + offset = 0;
>> + } else if (!s->got_profile && !s->need_profile) {
>> + for (int level = 0; level <= s->levels; level++)
>> + profile[level] = s->sigma / 1000000.;
>> + }
>> +
>> + for (int level = 0; level <= s->levels; level++) {
>> + const double ak = 2.0 + ((4.0 * level) / s->levels);
>> + const int length = s->length[level];
>> +
>> + power_average(wtc + offset, power + offset, length);
>> + noise_filter(power + offset, filter + offset, profile[level], ak,
>> length);
>> + signal_filter(power + offset, signal + offset, profile[level],
>> ak, length);
>> + denoise_level(wtc + offset, wtc + offset, filter + offset, signal
>> + offset, s->percent, length);
>> + offset += length;
>> + }
>> +
>> + idwt_levels(s, s->levels, dst, temp0, wtc);
>> +
>> + 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 = NULL;
>> +
>> + out = ff_get_audio_buffer(outlink, in->nb_samples);
>> + if (!out) {
>> + av_frame_free(&in);
>> + return AVERROR(ENOMEM);
>> + }
>> + out->pts = in->pts;
>> +
>> + td.in = in;
>> + td.out = out;
>> + ctx->internal->execute(ctx, filter_channel, &td, NULL,
>> inlink->channels);
>> + if (s->need_profile)
>> + s->got_profile = 1;
>> +
>> + av_frame_free(&in);
>> + return ff_filter_frame(outlink, out);
>> +}
>> +
>> +static int activate(AVFilterContext *ctx)
>> +{
>> + AVFilterLink *inlink = ctx->inputs[0];
>> + AVFilterLink *outlink = ctx->outputs[0];
>> + AudioFWTDNContext *s = ctx->priv;
>> + AVFrame *in = NULL;
>> + int ret;
>> +
>> + 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);
>> +
>> + FF_FILTER_FORWARD_STATUS(inlink, outlink);
>> + 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->power);
>> + av_frame_free(&s->profile);
>> + av_frame_free(&s->signal);
>> + av_frame_free(&s->temp0);
>> + av_frame_free(&s->temp1);
>> + av_frame_free(&s->wtc);
>> +}
>> +
>> +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("Reduce broadband noise from
>> stream using Fast Wavelet Transform."),
>> + .query_formats = query_formats,
>> + .priv_size = sizeof(AudioFWTDNContext),
>> + .priv_class = &afwtdn_class,
>> + .activate = activate,
>> + .uninit = uninit,
>> + .inputs = inputs,
>> + .outputs = outputs,
>> + .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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