/*
  * Copyright (c) 2018 The FFmpeg Project
  *
  * 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/audio_fifo.h"
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/opt.h"
 #include "libavcodec/avfft.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "formats.h"

 #include "filters.h"

 
 #define C       (M_LN10 * 0.1)
 #define RATIO    0.98
 #define RRATIO  (1.0 - RATIO)
 
 enum OutModes {
     IN_MODE,
     OUT_MODE,
     NOISE_MODE,
     NB_MODES
 };
 
 enum NoiseType {
     WHITE_NOISE,
     VINYL_NOISE,
     SHELLAC_NOISE,
     CUSTOM_NOISE,
     NB_NOISE
 };
 
 typedef struct DeNoiseChannel {
     int         band_noise[15];
     double      noise_band_auto_var[15];
     double      noise_band_sample[15];
     double     *amt;
     double     *band_amt;
     double     *band_excit;
     double     *gain;
     double     *prior;
     double     *prior_band_excit;
     double     *clean_data;
     double     *noisy_data;
     double     *out_samples;
     double     *spread_function;
     double     *abs_var;
     double     *rel_var;
     double     *min_abs_var;
     FFTComplex *fft_data;
     FFTContext *fft, *ifft;
 
     double      noise_band_norm[15];
     double      noise_band_avr[15];
     double      noise_band_avi[15];
     double      noise_band_var[15];
 
     double      sfm_threshold;
     double      sfm_alpha;
     double      sfm_results[3];
     int         sfm_fail_flags[512];
     int         sfm_fail_total;
 } DeNoiseChannel;
 
 typedef struct AudioFFTDeNoiseContext {
     const AVClass *class;
 
     float   noise_reduction;
     float   noise_floor;
     int     noise_type;
     char   *band_noise_str;
     float   residual_floor;
     int     track_noise;
     int     track_residual;
     int     output_mode;
 
     float   last_residual_floor;
     float   last_noise_floor;
     float   last_noise_reduction;
     float   last_noise_balance;
     int64_t block_count;
 
     int64_t pts;
     int     channels;
     int     sample_noise;
     int     sample_noise_start;
     int     sample_noise_end;
     float   sample_rate;
     int     buffer_length;
     int     fft_length;
     int     fft_length2;
     int     bin_count;
     int     window_length;
     int     sample_advance;
     int     number_of_bands;
 
     int     band_centre[15];
 
     int    *bin2band;
     double *window;
     double *band_alpha;
     double *band_beta;
 
     DeNoiseChannel *dnch;
 
     double  max_gain;
     double  max_var;
     double  gain_scale;
     double  window_weight;
     double  floor;
     double  sample_floor;
     double  auto_floor;
 
     int     noise_band_edge[17];
     int     noise_band_count;
     double  matrix_a[25];
     double  vector_b[5];
     double  matrix_b[75];
     double  matrix_c[75];
 
     AVAudioFifo *fifo;
 } AudioFFTDeNoiseContext;
 
 #define OFFSET(x) offsetof(AudioFFTDeNoiseContext, 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 afftdn_options[] = {

     { "nr", "set the noise reduction",    OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT,  {.dbl = 12},          .01, 97, AFR },
     { "nf", "set the noise floor",        OFFSET(noise_floor),     AV_OPT_TYPE_FLOAT,  {.dbl =-50},          -80,-20, AFR },
     { "nt", "set the noise type",         OFFSET(noise_type),      AV_OPT_TYPE_INT,    {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, "type" },
     {  "w", "white noise",                0,                       AV_OPT_TYPE_CONST,  {.i64 = WHITE_NOISE},   0,  0, AF, "type" },
     {  "v", "vinyl noise",                0,                       AV_OPT_TYPE_CONST,  {.i64 = VINYL_NOISE},   0,  0, AF, "type" },
     {  "s", "shellac noise",              0,                       AV_OPT_TYPE_CONST,  {.i64 = SHELLAC_NOISE}, 0,  0, AF, "type" },
     {  "c", "custom noise",               0,                       AV_OPT_TYPE_CONST,  {.i64 = CUSTOM_NOISE},  0,  0, AF, "type" },
     { "bn", "set the custom bands noise", OFFSET(band_noise_str),  AV_OPT_TYPE_STRING, {.str = 0},             0,  0, AF },
     { "rf", "set the residual floor",     OFFSET(residual_floor),  AV_OPT_TYPE_FLOAT,  {.dbl =-38},          -80,-20, AFR },
     { "tn", "track noise",                OFFSET(track_noise),     AV_OPT_TYPE_BOOL,   {.i64 =  0},            0,  1, AFR },
     { "tr", "track residual",             OFFSET(track_residual),  AV_OPT_TYPE_BOOL,   {.i64 =  0},            0,  1, AFR },
     { "om", "set output mode",            OFFSET(output_mode),     AV_OPT_TYPE_INT,    {.i64 = OUT_MODE},      0,  NB_MODES-1, AFR, "mode" },
     {  "i", "input",                      0,                       AV_OPT_TYPE_CONST,  {.i64 = IN_MODE},       0,  0, AFR, "mode" },
     {  "o", "output",                     0,                       AV_OPT_TYPE_CONST,  {.i64 = OUT_MODE},      0,  0, AFR, "mode" },
     {  "n", "noise",                      0,                       AV_OPT_TYPE_CONST,  {.i64 = NOISE_MODE},    0,  0, AFR, "mode" },

     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(afftdn);
 
 static int get_band_noise(AudioFFTDeNoiseContext *s,
                           int band, double a,
                           double b, double c)
 {
     double d1, d2, d3;
 
     d1 = a / s->band_centre[band];
     d1 = 10.0 * log(1.0 + d1 * d1) / M_LN10;
     d2 = b / s->band_centre[band];
     d2 = 10.0 * log(1.0 + d2 * d2) / M_LN10;
     d3 = s->band_centre[band] / c;
     d3 = 10.0 * log(1.0 + d3 * d3) / M_LN10;
 
     return lrint(-d1 + d2 - d3);
 }
 
 static void factor(double *array, int size)
 {
     for (int i = 0; i < size - 1; i++) {
         for (int j = i + 1; j < size; j++) {
             double d = array[j + i * size] / array[i + i * size];
 
             array[j + i * size] = d;
             for (int k = i + 1; k < size; k++) {
                 array[j + k * size] -= d * array[i + k * size];
             }
         }
     }
 }
 
 static void solve(double *matrix, double *vector, int size)
 {
     for (int i = 0; i < size - 1; i++) {
         for (int j = i + 1; j < size; j++) {
             double d = matrix[j + i * size];
             vector[j] -= d * vector[i];
         }
     }
 
     vector[size - 1] /= matrix[size * size - 1];
 
     for (int i = size - 2; i >= 0; i--) {
         double d = vector[i];
         for (int j = i + 1; j < size; j++)
             d -= matrix[i + j * size] * vector[j];
         vector[i] = d / matrix[i + i * size];
     }
 }
 
 static int process_get_band_noise(AudioFFTDeNoiseContext *s,
                                   DeNoiseChannel *dnch,
                                   int band)
 {
     double product, sum, f;
     int i = 0;
 
     if (band < 15)
         return dnch->band_noise[band];
 
     for (int j = 0; j < 5; j++) {
         sum = 0.0;
         for (int k = 0; k < 15; k++)
             sum += s->matrix_b[i++] * dnch->band_noise[k];
         s->vector_b[j] = sum;
     }
 
     solve(s->matrix_a, s->vector_b, 5);
     f = (0.5 * s->sample_rate) / s->band_centre[14];
     f = 15.0 + log(f / 1.5) / log(1.5);
     sum = 0.0;
     product = 1.0;
     for (int j = 0; j < 5; j++) {
         sum += product * s->vector_b[j];
         product *= f;
     }
 
     return lrint(sum);
 }
 
 static void calculate_sfm(AudioFFTDeNoiseContext *s,
                           DeNoiseChannel *dnch,
                           int start, int end)
 {
     double d1 = 0.0, d2 = 1.0;
     int i = 0, j = 0;
 
     for (int k = start; k < end; k++) {
         if (dnch->noisy_data[k] > s->sample_floor) {
             j++;
             d1 += dnch->noisy_data[k];
             d2 *= dnch->noisy_data[k];
             if (d2 > 1.0E100) {
                 d2 *= 1.0E-100;
                 i++;
             } else if (d2 < 1.0E-100) {
                 d2 *= 1.0E100;
                 i--;
             }
         }
     }
     if (j > 1) {
         d1 /= j;
         dnch->sfm_results[0] = d1;
         d2 = log(d2) + 230.2585 * i;
         d2 /= j;
         d1 = log(d1);
         dnch->sfm_results[1] = d1;
         dnch->sfm_results[2] = d1 - d2;
     } else {
         dnch->sfm_results[0] = s->auto_floor;
         dnch->sfm_results[1] = dnch->sfm_threshold;
         dnch->sfm_results[2] = dnch->sfm_threshold;
     }
 }
 
 static double limit_gain(double a, double b)
 {
     if (a > 1.0)
         return (b * a - 1.0) / (b + a - 2.0);
     if (a < 1.0)
         return (b * a - 2.0 * a + 1.0) / (b - a);
     return 1.0;
 }
 
 static void process_frame(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch,
                           FFTComplex *fft_data,
                           double *prior, double *prior_band_excit, int track_noise)
 {
     double d1, d2, d3, gain;
     int n, i1;
 
     d1 = fft_data[0].re * fft_data[0].re;
     dnch->noisy_data[0] = d1;
     d2 = d1 / dnch->abs_var[0];
     d3 = RATIO * prior[0] + RRATIO * fmax(d2 - 1.0, 0.0);
     gain = d3 / (1.0 + d3);
     gain *= (gain + M_PI_4 / fmax(d2, 1.0E-6));
     prior[0] = (d2 * gain);
     dnch->clean_data[0] = (d1 * gain);
     gain = sqrt(gain);
     dnch->gain[0] = gain;
     n = 0;
     for (int i = 1; i < s->fft_length2; i++) {
         d1 = fft_data[i].re * fft_data[i].re + fft_data[i].im * fft_data[i].im;
         if (d1 > s->sample_floor)
             n = i;
 
         dnch->noisy_data[i] = d1;
         d2 = d1 / dnch->abs_var[i];
         d3 = RATIO * prior[i] + RRATIO * fmax(d2 - 1.0, 0.0);
         gain = d3 / (1.0 + d3);
         gain *= (gain + M_PI_4 / fmax(d2, 1.0E-6));
         prior[i] = d2 * gain;
         dnch->clean_data[i] = d1 * gain;
         gain = sqrt(gain);
         dnch->gain[i] = gain;
     }
     d1 = fft_data[0].im * fft_data[0].im;
     if (d1 > s->sample_floor)
         n = s->fft_length2;
 
     dnch->noisy_data[s->fft_length2] = d1;
     d2 = d1 / dnch->abs_var[s->fft_length2];
     d3 = RATIO * prior[s->fft_length2] + RRATIO * fmax(d2 - 1.0, 0.0);
     gain = d3 / (1.0 + d3);
     gain *= gain + M_PI_4 / fmax(d2, 1.0E-6);
     prior[s->fft_length2] = d2 * gain;
     dnch->clean_data[s->fft_length2] = d1 * gain;
     gain = sqrt(gain);
     dnch->gain[s->fft_length2] = gain;
     if (n > s->fft_length2 - 2) {
         n = s->bin_count;
         i1 = s->noise_band_count;
     } else {
         i1 = 0;
         for (int i = 0; i <= s->noise_band_count; i++) {
             if (n > 1.1 * s->noise_band_edge[i]) {
                 i1 = i;
             }
         }
     }
 
     if (track_noise && (i1 > s->noise_band_count / 2)) {
         int j = FFMIN(n, s->noise_band_edge[i1]);
         int m = 3, k;
 
         for (k = i1 - 1; k >= 0; k--) {
             int i = s->noise_band_edge[k];
             calculate_sfm(s, dnch, i, j);
             dnch->noise_band_sample[k] = dnch->sfm_results[0];
             if (dnch->sfm_results[2] + 0.013 * m * fmax(0.0, dnch->sfm_results[1] - 20.53) >= dnch->sfm_threshold) {
                 break;
             }
             j = i;
             m++;
         }
 
         if (k < i1 - 1) {
             double sum = 0.0, min, max;
             int i;
 
             for (i = i1 - 1; i > k; i--) {
                 min = log(dnch->noise_band_sample[i] / dnch->noise_band_auto_var[i]);
                 sum += min;
             }
 
             i = i1 - k - 1;
             if (i < 5) {
                 min = 3.0E-4 * i * i;
             } else {
                 min = 3.0E-4 * (8 * i - 16);
             }
             if (i < 3) {
                 max = 2.0E-4 * i * i;
             } else {
                 max = 2.0E-4 * (4 * i - 4);
             }
 
             if (s->track_residual) {
                 if (s->last_noise_floor > s->last_residual_floor + 9) {
                     min *= 0.5;
                     max *= 0.75;
                 } else if (s->last_noise_floor > s->last_residual_floor + 6) {
                     min *= 0.4;
                     max *= 1.0;
                 } else if (s->last_noise_floor > s->last_residual_floor + 4) {
                     min *= 0.3;
                     max *= 1.3;
                 } else if (s->last_noise_floor > s->last_residual_floor + 2) {
                     min *= 0.2;
                     max *= 1.6;
                 } else if (s->last_noise_floor > s->last_residual_floor) {
                     min *= 0.1;
                     max *= 2.0;
                 } else {
                     min = 0.0;
                     max *= 2.5;
                 }
             }
 
             sum = av_clipd(sum, -min, max);
             sum = exp(sum);
             for (int i = 0; i < 15; i++)
                 dnch->noise_band_auto_var[i] *= sum;
         } else if (dnch->sfm_results[2] >= dnch->sfm_threshold) {
             dnch->sfm_fail_flags[s->block_count & 0x1FF] = 1;
             dnch->sfm_fail_total += 1;
         }
     }
 
     for (int i = 0; i < s->number_of_bands; i++) {
         dnch->band_excit[i] = 0.0;
         dnch->band_amt[i] = 0.0;
     }
 
     for (int i = 0; i < s->bin_count; i++) {
         dnch->band_excit[s->bin2band[i]] += dnch->clean_data[i];
     }
 
     for (int i = 0; i < s->number_of_bands; i++) {
         dnch->band_excit[i] = fmax(dnch->band_excit[i],
                                 s->band_alpha[i] * dnch->band_excit[i] +
                                 s->band_beta[i] * prior_band_excit[i]);
         prior_band_excit[i] = dnch->band_excit[i];
     }
 
     for (int j = 0, i = 0; j < s->number_of_bands; j++) {
         for (int k = 0; k < s->number_of_bands; k++) {
             dnch->band_amt[j] += dnch->spread_function[i++] * dnch->band_excit[k];
         }
     }
 
     for (int i = 0; i < s->bin_count; i++)
         dnch->amt[i] = dnch->band_amt[s->bin2band[i]];
 
     if (dnch->amt[0] > dnch->abs_var[0]) {
         dnch->gain[0] = 1.0;
     } else if (dnch->amt[0] > dnch->min_abs_var[0]) {
         double limit = sqrt(dnch->abs_var[0] / dnch->amt[0]);
         dnch->gain[0] = limit_gain(dnch->gain[0], limit);
     } else {
         dnch->gain[0] = limit_gain(dnch->gain[0], s->max_gain);
     }
     if (dnch->amt[s->fft_length2] > dnch->abs_var[s->fft_length2]) {
         dnch->gain[s->fft_length2] = 1.0;
     } else if (dnch->amt[s->fft_length2] > dnch->min_abs_var[s->fft_length2]) {
         double limit = sqrt(dnch->abs_var[s->fft_length2] / dnch->amt[s->fft_length2]);
         dnch->gain[s->fft_length2] = limit_gain(dnch->gain[s->fft_length2], limit);
     } else {
         dnch->gain[s->fft_length2] = limit_gain(dnch->gain[s->fft_length2], s->max_gain);
     }
 
     for (int i = 1; i < s->fft_length2; i++) {
         if (dnch->amt[i] > dnch->abs_var[i]) {
             dnch->gain[i] = 1.0;
         } else if (dnch->amt[i] > dnch->min_abs_var[i]) {
             double limit = sqrt(dnch->abs_var[i] / dnch->amt[i]);
             dnch->gain[i] = limit_gain(dnch->gain[i], limit);
         } else {
             dnch->gain[i] = limit_gain(dnch->gain[i], s->max_gain);
         }
     }
 
     gain = dnch->gain[0];
     dnch->clean_data[0] = (gain * gain * dnch->noisy_data[0]);
     fft_data[0].re *= gain;
     gain = dnch->gain[s->fft_length2];
     dnch->clean_data[s->fft_length2] = (gain * gain * dnch->noisy_data[s->fft_length2]);
     fft_data[0].im *= gain;
     for (int i = 1; i < s->fft_length2; i++) {
         gain = dnch->gain[i];
         dnch->clean_data[i] = (gain * gain * dnch->noisy_data[i]);
         fft_data[i].re *= gain;
         fft_data[i].im *= gain;
     }
 }
 
 static double freq2bark(double x)
 {
     double d = x / 7500.0;
 
     return 13.0 * atan(7.6E-4 * x) + 3.5 * atan(d * d);
 }
 
 static int get_band_centre(AudioFFTDeNoiseContext *s, int band)
 {
     if (band == -1)
         return lrint(s->band_centre[0] / 1.5);
 
     return s->band_centre[band];
 }
 
 static int get_band_edge(AudioFFTDeNoiseContext *s, int band)
 {
     int i;
 
     if (band == 15) {
         i = lrint(s->band_centre[14] * 1.224745);
     } else {
         i = lrint(s->band_centre[band] / 1.224745);
     }
 
     return FFMIN(i, s->sample_rate / 2);
 }
 
 static void set_band_parameters(AudioFFTDeNoiseContext *s,
                                 DeNoiseChannel *dnch)
 {
     double band_noise, d2, d3, d4, d5;
     int i = 0, j = 0, k = 0;
 
     d5 = 0.0;
     band_noise = process_get_band_noise(s, dnch, 0);
     for (int m = j; m <= s->fft_length2; m++) {
         if (m == j) {
             i = j;
             d5 = band_noise;
             if (k == 15) {
                 j = s->bin_count;
             } else {
                 j = s->fft_length * get_band_centre(s, k) / s->sample_rate;
             }
             d2 = j - i;
             band_noise = process_get_band_noise(s, dnch, k);
             k++;
         }
         d3 = (j - m) / d2;
         d4 = (m - i) / d2;
         dnch->rel_var[m] = exp((d5 * d3 + band_noise * d4) * C);
     }
     dnch->rel_var[s->fft_length2] = exp(band_noise * C);
 
     for (i = 0; i < 15; i++)
         dnch->noise_band_auto_var[i] = s->max_var * exp((process_get_band_noise(s, dnch, i) - 2.0) * C);
 
     for (i = 0; i <= s->fft_length2; i++) {
         dnch->abs_var[i] = fmax(s->max_var * dnch->rel_var[i], 1.0);
         dnch->min_abs_var[i] = s->gain_scale * dnch->abs_var[i];
     }
 }
 
 static void read_custom_noise(AudioFFTDeNoiseContext *s, int ch)
 {
     DeNoiseChannel *dnch = &s->dnch[ch];
     char *p, *arg, *saveptr = NULL;
     int i, ret, band_noise[15] = { 0 };
 
     if (!s->band_noise_str)
         return;
 
     p = av_strdup(s->band_noise_str);
     if (!p)
         return;
 
     for (i = 0; i < 15; i++) {
         if (!(arg = av_strtok(p, "| ", &saveptr)))
             break;
 
         p = NULL;
 

         ret = av_sscanf(arg, "%d", &band_noise[i]);

         if (ret != 1) {
             av_log(s, AV_LOG_ERROR, "Custom band noise must be integer.\n");
             break;
         }
 
         band_noise[i] = av_clip(band_noise[i], -24, 24);
     }
 
     av_free(p);
     memcpy(dnch->band_noise, band_noise, sizeof(band_noise));
 }
 
 static void set_parameters(AudioFFTDeNoiseContext *s)
 {
     if (s->last_noise_floor != s->noise_floor)
         s->last_noise_floor = s->noise_floor;
 
     if (s->track_residual)
         s->last_noise_floor = fmaxf(s->last_noise_floor, s->residual_floor);
 
     s->max_var = s->floor * exp((100.0 + s->last_noise_floor) * C);
 
     if (s->track_residual) {
         s->last_residual_floor = s->residual_floor;
         s->last_noise_reduction = fmax(s->last_noise_floor - s->last_residual_floor, 0);
         s->max_gain = exp(s->last_noise_reduction * (0.5 * C));
     } else if (s->noise_reduction != s->last_noise_reduction) {
         s->last_noise_reduction = s->noise_reduction;
         s->last_residual_floor = av_clipf(s->last_noise_floor - s->last_noise_reduction, -80, -20);
         s->max_gain = exp(s->last_noise_reduction * (0.5 * C));
     }
 
     s->gain_scale = 1.0 / (s->max_gain * s->max_gain);
 
     for (int ch = 0; ch < s->channels; ch++) {
         DeNoiseChannel *dnch = &s->dnch[ch];
 
         set_band_parameters(s, dnch);
     }
 }
 
 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioFFTDeNoiseContext *s = ctx->priv;
     double wscale, sar, sum, sdiv;
     int i, j, k, m, n;
 
     s->dnch = av_calloc(inlink->channels, sizeof(*s->dnch));
     if (!s->dnch)
         return AVERROR(ENOMEM);
 
     s->pts = AV_NOPTS_VALUE;
     s->channels = inlink->channels;
     s->sample_rate = inlink->sample_rate;
     s->sample_advance = s->sample_rate / 80;
     s->window_length = 3 * s->sample_advance;
     s->fft_length2 = 1 << (32 - ff_clz(s->window_length));
     s->fft_length = s->fft_length2 * 2;
     s->buffer_length = s->fft_length * 2;
     s->bin_count = s->fft_length2 + 1;
 
     s->band_centre[0] = 80;
     for (i = 1; i < 15; i++) {
         s->band_centre[i] = lrint(1.5 * s->band_centre[i - 1] + 5.0);
         if (s->band_centre[i] < 1000) {
             s->band_centre[i] = 10 * (s->band_centre[i] / 10);
         } else if (s->band_centre[i] < 5000) {
             s->band_centre[i] = 50 * ((s->band_centre[i] + 20) / 50);
         } else if (s->band_centre[i] < 15000) {
             s->band_centre[i] = 100 * ((s->band_centre[i] + 45) / 100);
         } else {
             s->band_centre[i] = 1000 * ((s->band_centre[i] + 495) / 1000);
         }
     }
 
     for (j = 0; j < 5; j++) {
         for (k = 0; k < 5; k++) {
             s->matrix_a[j + k * 5] = 0.0;
             for (m = 0; m < 15; m++)
                 s->matrix_a[j + k * 5] += pow(m, j + k);
         }
     }
 
     factor(s->matrix_a, 5);
 
     i = 0;
     for (j = 0; j < 5; j++)
         for (k = 0; k < 15; k++)
             s->matrix_b[i++] = pow(k, j);
 
     i = 0;
     for (j = 0; j < 15; j++)
         for (k = 0; k < 5; k++)
             s->matrix_c[i++] = pow(j, k);
 
     s->window = av_calloc(s->window_length, sizeof(*s->window));
     s->bin2band = av_calloc(s->bin_count, sizeof(*s->bin2band));
     if (!s->window || !s->bin2band)
         return AVERROR(ENOMEM);
 
     sdiv = s->sample_rate / 17640.0;
     for (i = 0; i <= s->fft_length2; i++)
         s->bin2band[i] = lrint(sdiv * freq2bark((0.5 * i * s->sample_rate) / s->fft_length2));
 
     s->number_of_bands = s->bin2band[s->fft_length2] + 1;
 
     s->band_alpha = av_calloc(s->number_of_bands, sizeof(*s->band_alpha));
     s->band_beta = av_calloc(s->number_of_bands, sizeof(*s->band_beta));
     if (!s->band_alpha || !s->band_beta)
         return AVERROR(ENOMEM);
 
     for (int ch = 0; ch < inlink->channels; ch++) {
         DeNoiseChannel *dnch = &s->dnch[ch];
 
         switch (s->noise_type) {
         case WHITE_NOISE:
             for (i = 0; i < 15; i++)
                 dnch->band_noise[i] = 0;
             break;
         case VINYL_NOISE:
             for (i = 0; i < 15; i++)
                 dnch->band_noise[i] = get_band_noise(s, i, 50.0, 500.5, 2125.0) + FFMAX(i - 7, 0);
             break;
         case SHELLAC_NOISE:
             for (i = 0; i < 15; i++)
                 dnch->band_noise[i] = get_band_noise(s, i, 1.0, 500.0, 1.0E10) + FFMAX(i - 12, -5);

             break;

         case CUSTOM_NOISE:
             read_custom_noise(s, ch);
             break;
         default:
             return AVERROR_BUG;
         }
 
 
         dnch->sfm_threshold = 0.8;
         dnch->sfm_alpha = 0.05;
         for (i = 0; i < 512; i++)
             dnch->sfm_fail_flags[i] = 0;
 
         dnch->sfm_fail_total = 0;
         j = FFMAX((int)(10.0 * (1.3 - dnch->sfm_threshold)), 1);
 
         for (i = 0; i < 512; i += j) {
             dnch->sfm_fail_flags[i] = 1;
             dnch->sfm_fail_total += 1;
         }
 
         dnch->amt = av_calloc(s->bin_count, sizeof(*dnch->amt));
         dnch->band_amt = av_calloc(s->number_of_bands, sizeof(*dnch->band_amt));
         dnch->band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->band_excit));
         dnch->gain = av_calloc(s->bin_count, sizeof(*dnch->gain));
         dnch->prior = av_calloc(s->bin_count, sizeof(*dnch->prior));
         dnch->prior_band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->prior_band_excit));
         dnch->clean_data = av_calloc(s->bin_count, sizeof(*dnch->clean_data));
         dnch->noisy_data = av_calloc(s->bin_count, sizeof(*dnch->noisy_data));
         dnch->out_samples = av_calloc(s->buffer_length, sizeof(*dnch->out_samples));
         dnch->abs_var = av_calloc(s->bin_count, sizeof(*dnch->abs_var));
         dnch->rel_var = av_calloc(s->bin_count, sizeof(*dnch->rel_var));
         dnch->min_abs_var = av_calloc(s->bin_count, sizeof(*dnch->min_abs_var));
         dnch->fft_data = av_calloc(s->fft_length2 + 1, sizeof(*dnch->fft_data));
         dnch->fft  = av_fft_init(av_log2(s->fft_length2), 0);
         dnch->ifft = av_fft_init(av_log2(s->fft_length2), 1);
         dnch->spread_function = av_calloc(s->number_of_bands * s->number_of_bands,
                                           sizeof(*dnch->spread_function));
 
         if (!dnch->amt ||
             !dnch->band_amt ||
             !dnch->band_excit ||
             !dnch->gain ||
             !dnch->prior ||
             !dnch->prior_band_excit ||
             !dnch->clean_data ||
             !dnch->noisy_data ||
             !dnch->out_samples ||
             !dnch->fft_data ||
             !dnch->abs_var ||
             !dnch->rel_var ||
             !dnch->min_abs_var ||
             !dnch->spread_function ||
             !dnch->fft ||
             !dnch->ifft)
             return AVERROR(ENOMEM);
     }
 
     for (int ch = 0; ch < inlink->channels; ch++) {
         DeNoiseChannel *dnch = &s->dnch[ch];
         double *prior_band_excit = dnch->prior_band_excit;
         double *prior = dnch->prior;
         double min, max;
         double p1, p2;
 
         p1 = pow(0.1, 2.5 / sdiv);
         p2 = pow(0.1, 1.0 / sdiv);
         j = 0;
         for (m = 0; m < s->number_of_bands; m++) {
             for (n = 0; n < s->number_of_bands; n++) {
                 if (n < m) {
                     dnch->spread_function[j++] = pow(p2, m - n);
                 } else if (n > m) {
                     dnch->spread_function[j++] = pow(p1, n - m);
                 } else {
                     dnch->spread_function[j++] = 1.0;
                 }
             }
         }
 
         for (m = 0; m < s->number_of_bands; m++) {
             dnch->band_excit[m] = 0.0;
             prior_band_excit[m] = 0.0;
         }
 
         for (m = 0; m <= s->fft_length2; m++)
             dnch->band_excit[s->bin2band[m]] += 1.0;
 
         j = 0;
         for (m = 0; m < s->number_of_bands; m++) {
             for (n = 0; n < s->number_of_bands; n++)
                 prior_band_excit[m] += dnch->spread_function[j++] * dnch->band_excit[n];
         }
 
         min = pow(0.1, 2.5);
         max = pow(0.1, 1.0);
         for (int i = 0; i < s->number_of_bands; i++) {
             if (i < lrint(12.0 * sdiv)) {
                 dnch->band_excit[i] = pow(0.1, 1.45 + 0.1 * i / sdiv);
             } else {
                 dnch->band_excit[i] = pow(0.1, 2.5 - 0.2 * (i / sdiv - 14.0));
             }
             dnch->band_excit[i] = av_clipd(dnch->band_excit[i], min, max);
         }
 
         for (int i = 0; i <= s->fft_length2; i++)
             prior[i] = RRATIO;
         for (int i = 0; i < s->buffer_length; i++)
             dnch->out_samples[i] = 0;
 
         j = 0;
         for (int i = 0; i < s->number_of_bands; i++)
             for (int k = 0; k < s->number_of_bands; k++)
                 dnch->spread_function[j++] *= dnch->band_excit[i] / prior_band_excit[i];
     }
 
     j = 0;
     sar = s->sample_advance / s->sample_rate;
     for (int i = 0; i <= s->fft_length2; i++) {
         if ((i == s->fft_length2) || (s->bin2band[i] > j)) {
             double d6 = (i - 1) * s->sample_rate / s->fft_length;
             double d7 = fmin(0.008 + 2.2 / d6, 0.03);
             s->band_alpha[j] = exp(-sar / d7);
             s->band_beta[j] = 1.0 - s->band_alpha[j];
             j = s->bin2band[i];
         }
     }
 
     wscale = sqrt(16.0 / (9.0 * s->fft_length));
     sum = 0.0;
     for (int i = 0; i < s->window_length; i++) {
         double d10 = sin(i * M_PI / s->window_length);
         d10 *= wscale * d10;
         s->window[i] = d10;
         sum += d10 * d10;
     }
 
     s->window_weight = 0.5 * sum;
     s->floor = (1LL << 48) * exp(-23.025558369790467) * s->window_weight;
     s->sample_floor = s->floor * exp(4.144600506562284);
     s->auto_floor = s->floor * exp(6.907667510937141);
 
     set_parameters(s);
 
     s->noise_band_edge[0] = FFMIN(s->fft_length2, s->fft_length * get_band_edge(s, 0) / s->sample_rate);
     i = 0;
     for (int j = 1; j < 16; j++) {
         s->noise_band_edge[j] = FFMIN(s->fft_length2, s->fft_length * get_band_edge(s, j) / s->sample_rate);
         if (s->noise_band_edge[j] > lrint(1.1 * s->noise_band_edge[j - 1]))
             i++;
         s->noise_band_edge[16] = i;
     }
     s->noise_band_count = s->noise_band_edge[16];
 
     s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->fft_length);
     if (!s->fifo)
         return AVERROR(ENOMEM);
 
     return 0;
 }
 
 static void preprocess(FFTComplex *in, int len)
 {
     double d1, d2, d3, d4, d5, d6, d7, d8, d9, d10;
     int n, i, k;
 
     d5 = 2.0 * M_PI / len;
     d8 = sin(0.5 * d5);
     d8 = -2.0 * d8 * d8;
     d7 = sin(d5);
     d9 = 1.0 + d8;
     d6 = d7;
     n = len / 2;
 
     for (i = 1; i < len / 4; i++) {
         k = n - i;
         d2 = 0.5 * (in[i].re + in[k].re);
         d1 = 0.5 * (in[i].im - in[k].im);
         d4 = 0.5 * (in[i].im + in[k].im);
         d3 = 0.5 * (in[k].re - in[i].re);
         in[i].re = d2 + d9 * d4 + d6 * d3;
         in[i].im = d1 + d9 * d3 - d6 * d4;
         in[k].re = d2 - d9 * d4 - d6 * d3;
         in[k].im = -d1 + d9 * d3 - d6 * d4;
         d10 = d9;
         d9 += d9 * d8 - d6 * d7;
         d6 += d6 * d8 + d10 * d7;
     }
 
     d2 = in[0].re;
     in[0].re = d2 + in[0].im;
     in[0].im = d2 - in[0].im;
 }
 
 static void postprocess(FFTComplex *in, int len)
 {
     double d1, d2, d3, d4, d5, d6, d7, d8, d9, d10;
     int n, i, k;
 
     d5 = 2.0 * M_PI / len;
     d8 = sin(0.5 * d5);
     d8 = -2.0 * d8 * d8;
     d7 = sin(d5);
     d9 = 1.0 + d8;
     d6 = d7;
     n = len / 2;
     for (i = 1; i < len / 4; i++) {
         k = n - i;
         d2 = 0.5 * (in[i].re + in[k].re);
         d1 = 0.5 * (in[i].im - in[k].im);
         d4 = 0.5 * (in[i].re - in[k].re);
         d3 = 0.5 * (in[i].im + in[k].im);
         in[i].re = d2 - d9 * d3 - d6 * d4;
         in[i].im = d1 + d9 * d4 - d6 * d3;
         in[k].re = d2 + d9 * d3 + d6 * d4;
         in[k].im = -d1 + d9 * d4 - d6 * d3;
         d10 = d9;
         d9 += d9 * d8 - d6 * d7;
         d6 += d6 * d8 + d10 * d7;
     }
     d2 = in[0].re;
     in[0].re = 0.5 * (d2 + in[0].im);
     in[0].im = 0.5 * (d2 - in[0].im);
 }
 
 static void init_sample_noise(DeNoiseChannel *dnch)
 {
     for (int i = 0; i < 15; i++) {
         dnch->noise_band_norm[i] = 0.0;
         dnch->noise_band_avr[i] = 0.0;
         dnch->noise_band_avi[i] = 0.0;
         dnch->noise_band_var[i] = 0.0;
     }
 }
 
 static void sample_noise_block(AudioFFTDeNoiseContext *s,
                                DeNoiseChannel *dnch,
                                AVFrame *in, int ch)
 {
     float *src = (float *)in->extended_data[ch];
     double mag2, var = 0.0, avr = 0.0, avi = 0.0;
     int edge, j, k, n, edgemax;
 
     for (int i = 0; i < s->window_length; i++) {
         dnch->fft_data[i].re = s->window[i] * src[i] * (1LL << 24);
         dnch->fft_data[i].im = 0.0;
     }
 
     for (int i = s->window_length; i < s->fft_length2; i++) {
         dnch->fft_data[i].re = 0.0;
         dnch->fft_data[i].im = 0.0;
     }
 
     av_fft_permute(dnch->fft, dnch->fft_data);
     av_fft_calc(dnch->fft, dnch->fft_data);
 
     preprocess(dnch->fft_data, s->fft_length);
 
     edge = s->noise_band_edge[0];
     j = edge;
     k = 0;
     n = j;
     edgemax = fmin(s->fft_length2, s->noise_band_edge[15]);
     dnch->fft_data[s->fft_length2].re = dnch->fft_data[0].im;
     dnch->fft_data[0].im = 0.0;
     dnch->fft_data[s->fft_length2].im = 0.0;
 
     for (int i = j; i <= edgemax; i++) {
         if ((i == j) && (i < edgemax)) {
             if (j > edge) {
                 dnch->noise_band_norm[k - 1] += j - edge;
                 dnch->noise_band_avr[k - 1] += avr;
                 dnch->noise_band_avi[k - 1] += avi;
                 dnch->noise_band_var[k - 1] += var;
             }
             k++;
             edge = j;
             j = s->noise_band_edge[k];
             if (k == 15) {
                 j++;
             }
             var = 0.0;
             avr = 0.0;
             avi = 0.0;
         }
         avr += dnch->fft_data[n].re;
         avi += dnch->fft_data[n].im;
         mag2 = dnch->fft_data[n].re * dnch->fft_data[n].re +
                dnch->fft_data[n].im * dnch->fft_data[n].im;
 
         mag2 = fmax(mag2, s->sample_floor);
 
         dnch->noisy_data[i] = mag2;
         var += mag2;
         n++;
     }
 
     dnch->noise_band_norm[k - 1] += j - edge;
     dnch->noise_band_avr[k - 1] += avr;
     dnch->noise_band_avi[k - 1] += avi;
     dnch->noise_band_var[k - 1] += var;
 }
 
 static void finish_sample_noise(AudioFFTDeNoiseContext *s,
                                 DeNoiseChannel *dnch,
                                 double *sample_noise)
 {
     for (int i = 0; i < s->noise_band_count; i++) {
         dnch->noise_band_avr[i] /= dnch->noise_band_norm[i];
         dnch->noise_band_avi[i] /= dnch->noise_band_norm[i];
         dnch->noise_band_var[i] /= dnch->noise_band_norm[i];
         dnch->noise_band_var[i] -= dnch->noise_band_avr[i] * dnch->noise_band_avr[i] +
                                    dnch->noise_band_avi[i] * dnch->noise_band_avi[i];
         dnch->noise_band_auto_var[i] = dnch->noise_band_var[i];
         sample_noise[i] = (1.0 / C) * log(dnch->noise_band_var[i] / s->floor) - 100.0;
     }
     if (s->noise_band_count < 15) {
         for (int i = s->noise_band_count; i < 15; i++)
             sample_noise[i] = sample_noise[i - 1];
     }
 }
 
 static void set_noise_profile(AudioFFTDeNoiseContext *s,
                               DeNoiseChannel *dnch,
                               double *sample_noise,
                               int new_profile)
 {
     int new_band_noise[15];
     double temp[15];
     double sum = 0.0, d1;
     float new_noise_floor;
     int i, n;
 
     for (int m = 0; m < 15; m++)
         temp[m] = sample_noise[m];
 
     if (new_profile) {
         i = 0;
         for (int m = 0; m < 5; m++) {
             sum = 0.0;
             for (n = 0; n < 15; n++)
                 sum += s->matrix_b[i++] * temp[n];
             s->vector_b[m] = sum;
         }
         solve(s->matrix_a, s->vector_b, 5);
         i = 0;
         for (int m = 0; m < 15; m++) {
             sum = 0.0;
             for (n = 0; n < 5; n++)
                 sum += s->matrix_c[i++] * s->vector_b[n];
             temp[m] = sum;
         }
     }
 
     sum = 0.0;
     for (int m = 0; m < 15; m++)
         sum += temp[m];
 
     d1 = (int)(sum / 15.0 - 0.5);
     if (!new_profile)
         i = lrint(temp[7] - d1);
 
     for (d1 -= dnch->band_noise[7] - i; d1 > -20.0; d1 -= 1.0)
         ;
 
     for (int m = 0; m < 15; m++)
         temp[m] -= d1;
 
     new_noise_floor = d1 + 2.5;
 
     if (new_profile) {
         av_log(s, AV_LOG_INFO, "bn=");
         for (int m = 0; m < 15; m++) {
             new_band_noise[m] = lrint(temp[m]);
             new_band_noise[m] = av_clip(new_band_noise[m], -24, 24);
             av_log(s, AV_LOG_INFO, "%d ", new_band_noise[m]);
         }
         av_log(s, AV_LOG_INFO, "\n");
         memcpy(dnch->band_noise, new_band_noise, sizeof(new_band_noise));
     }
 
     if (s->track_noise)
         s->noise_floor = new_noise_floor;
 }
 
 typedef struct ThreadData {
     AVFrame *in;
 } ThreadData;
 
 static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
 {
     AudioFFTDeNoiseContext *s = ctx->priv;
     ThreadData *td = arg;
     AVFrame *in = td->in;
     const int start = (in->channels * jobnr) / nb_jobs;
     const int end = (in->channels * (jobnr+1)) / nb_jobs;
 
     for (int ch = start; ch < end; ch++) {
         DeNoiseChannel *dnch = &s->dnch[ch];
         const float *src = (const float *)in->extended_data[ch];
         double *dst = dnch->out_samples;
 
         if (s->track_noise) {
             int i = s->block_count & 0x1FF;
 
             if (dnch->sfm_fail_flags[i])
                 dnch->sfm_fail_total--;
             dnch->sfm_fail_flags[i] = 0;
             dnch->sfm_threshold *= 1.0 - dnch->sfm_alpha;
             dnch->sfm_threshold += dnch->sfm_alpha * (0.5 + (1.0 / 640) * dnch->sfm_fail_total);
         }
 
         for (int m = 0; m < s->window_length; m++) {
             dnch->fft_data[m].re = s->window[m] * src[m] * (1LL << 24);
             dnch->fft_data[m].im = 0;
         }
 
         for (int m = s->window_length; m < s->fft_length2; m++) {
             dnch->fft_data[m].re = 0;
             dnch->fft_data[m].im = 0;
         }
 
         av_fft_permute(dnch->fft, dnch->fft_data);
         av_fft_calc(dnch->fft, dnch->fft_data);
 
         preprocess(dnch->fft_data, s->fft_length);
         process_frame(s, dnch, dnch->fft_data,
                       dnch->prior,
                       dnch->prior_band_excit,
                       s->track_noise);
         postprocess(dnch->fft_data, s->fft_length);
 
         av_fft_permute(dnch->ifft, dnch->fft_data);
         av_fft_calc(dnch->ifft, dnch->fft_data);
 
         for (int m = 0; m < s->window_length; m++)
             dst[m] += s->window[m] * dnch->fft_data[m].re / (1LL << 24);
     }
 
     return 0;
 }
 
 static void get_auto_noise_levels(AudioFFTDeNoiseContext *s,
                                   DeNoiseChannel *dnch,
                                   double *levels)
 {
     if (s->noise_band_count > 0) {
         for (int i = 0; i < s->noise_band_count; i++) {
             levels[i] = (1.0 / C) * log(dnch->noise_band_auto_var[i] / s->floor) - 100.0;
         }
         if (s->noise_band_count < 15) {
             for (int i = s->noise_band_count; i < 15; i++)
                 levels[i] = levels[i - 1];
         }
     } else {
         for (int i = 0; i < 15; i++) {
             levels[i] = -100.0;
         }
     }
 }
 

 static int output_frame(AVFilterLink *inlink)

 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     AudioFFTDeNoiseContext *s = ctx->priv;
     AVFrame *out = NULL, *in = NULL;
     ThreadData td;
     int ret = 0;
 

     in = ff_get_audio_buffer(outlink, s->window_length);
     if (!in)
         return AVERROR(ENOMEM);

     ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length);

     if (ret < 0)

         goto end;

     if (s->track_noise) {
         for (int ch = 0; ch < inlink->channels; ch++) {
             DeNoiseChannel *dnch = &s->dnch[ch];
             double levels[15];

             get_auto_noise_levels(s, dnch, levels);
             set_noise_profile(s, dnch, levels, 0);
         }

         if (s->noise_floor != s->last_noise_floor)
             set_parameters(s);
     }

     if (s->sample_noise_start) {
         for (int ch = 0; ch < inlink->channels; ch++) {
             DeNoiseChannel *dnch = &s->dnch[ch];

             init_sample_noise(dnch);

         }

         s->sample_noise_start = 0;
         s->sample_noise = 1;
     }

     if (s->sample_noise) {
         for (int ch = 0; ch < inlink->channels; ch++) {
             DeNoiseChannel *dnch = &s->dnch[ch];

             sample_noise_block(s, dnch, in, ch);

         }

     }

     if (s->sample_noise_end) {
         for (int ch = 0; ch < inlink->channels; ch++) {
             DeNoiseChannel *dnch = &s->dnch[ch];
             double sample_noise[15];

             finish_sample_noise(s, dnch, sample_noise);
             set_noise_profile(s, dnch, sample_noise, 1);
             set_band_parameters(s, dnch);

         }

         s->sample_noise = 0;
         s->sample_noise_end = 0;
     }

     s->block_count++;
     td.in = in;
     ctx->internal->execute(ctx, filter_channel, &td, NULL,
                            FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));

     out = ff_get_audio_buffer(outlink, s->sample_advance);
     if (!out) {
         ret = AVERROR(ENOMEM);
         goto end;
     }

     for (int ch = 0; ch < inlink->channels; ch++) {
         DeNoiseChannel *dnch = &s->dnch[ch];
         double *src = dnch->out_samples;
         float *orig = (float *)in->extended_data[ch];
         float *dst = (float *)out->extended_data[ch];
 
         switch (s->output_mode) {
         case IN_MODE:
             for (int m = 0; m < s->sample_advance; m++)
                 dst[m] = orig[m];
             break;
         case OUT_MODE:
             for (int m = 0; m < s->sample_advance; m++)
                 dst[m] = src[m];
             break;
         case NOISE_MODE:
             for (int m = 0; m < s->sample_advance; m++)
                 dst[m] = orig[m] - src[m];

             break;

         default:

             av_frame_free(&out);
             ret = AVERROR_BUG;
             goto end;

         }

         memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src));
         memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
     }

     av_audio_fifo_drain(s->fifo, s->sample_advance);
 
     out->pts = s->pts;
     ret = ff_filter_frame(outlink, out);
     if (ret < 0)
         goto end;

     s->pts += av_rescale_q(s->sample_advance, (AVRational){1, outlink->sample_rate}, outlink->time_base);

 end:
     av_frame_free(&in);

     return ret;
 }

 static int activate(AVFilterContext *ctx)
 {
     AVFilterLink *inlink = ctx->inputs[0];
     AVFilterLink *outlink = ctx->outputs[0];
     AudioFFTDeNoiseContext *s = ctx->priv;
     AVFrame *frame = NULL;
     int ret;
 
     FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
 
     ret = ff_inlink_consume_frame(inlink, &frame);
     if (ret < 0)
         return ret;
 
     if (ret > 0) {
         if (s->pts == AV_NOPTS_VALUE)
             s->pts = frame->pts;
 
         ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
         av_frame_free(&frame);

         if (ret < 0)

             return ret;

     }
 

     if (av_audio_fifo_size(s->fifo) >= s->window_length)
         return output_frame(inlink);
 
     FF_FILTER_FORWARD_STATUS(inlink, outlink);
     if (ff_outlink_frame_wanted(outlink) &&
         av_audio_fifo_size(s->fifo) < s->window_length) {
         ff_inlink_request_frame(inlink);
         return 0;
     }
 
     return FFERROR_NOT_READY;

 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioFFTDeNoiseContext *s = ctx->priv;
 
     av_freep(&s->window);
     av_freep(&s->bin2band);
     av_freep(&s->band_alpha);
     av_freep(&s->band_beta);
 
     if (s->dnch) {
         for (int ch = 0; ch < s->channels; ch++) {
             DeNoiseChannel *dnch = &s->dnch[ch];
             av_freep(&dnch->amt);
             av_freep(&dnch->band_amt);
             av_freep(&dnch->band_excit);
             av_freep(&dnch->gain);
             av_freep(&dnch->prior);
             av_freep(&dnch->prior_band_excit);
             av_freep(&dnch->clean_data);
             av_freep(&dnch->noisy_data);
             av_freep(&dnch->out_samples);
             av_freep(&dnch->spread_function);
             av_freep(&dnch->abs_var);
             av_freep(&dnch->rel_var);
             av_freep(&dnch->min_abs_var);
             av_freep(&dnch->fft_data);
             av_fft_end(dnch->fft);
             dnch->fft = NULL;
             av_fft_end(dnch->ifft);
             dnch->ifft = NULL;
         }
         av_freep(&s->dnch);
     }
 
     av_audio_fifo_free(s->fifo);
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats = NULL;
     AVFilterChannelLayouts *layouts = NULL;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_FLTP,
         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 process_command(AVFilterContext *ctx, const char *cmd, const char *args,
                            char *res, int res_len, int flags)
 {
     AudioFFTDeNoiseContext *s = ctx->priv;
     int need_reset = 0;

     int ret = 0;

 
     if (!strcmp(cmd, "sample_noise") ||
         !strcmp(cmd, "sn")) {
         if (!strcmp(args, "start")) {
             s->sample_noise_start = 1;
             s->sample_noise_end = 0;

         } else if (!strcmp(args, "end") ||
                    !strcmp(args, "stop")) {

             s->sample_noise_start = 0;
             s->sample_noise_end = 1;
         }

     } else {
         ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
         if (ret < 0)
             return ret;
         need_reset = 1;

     }
 
     if (need_reset)
         set_parameters(s);
 
     return 0;
 }
 
 static const AVFilterPad inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
     },
     { NULL }
 };
 
 static const AVFilterPad outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 
 AVFilter ff_af_afftdn = {
     .name            = "afftdn",
     .description     = NULL_IF_CONFIG_SMALL("Denoise audio samples using FFT."),
     .query_formats   = query_formats,
     .priv_size       = sizeof(AudioFFTDeNoiseContext),
     .priv_class      = &afftdn_class,

     .activate        = activate,

     .uninit          = uninit,
     .inputs          = inputs,
     .outputs         = outputs,
     .process_command = process_command,
     .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
                        AVFILTER_FLAG_SLICE_THREADS,
 };