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/*
* 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/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#define C (M_LN10 * 0.1)
#define SOLVE_SIZE (5)
#define NB_PROFILE_BANDS (15)
enum SampleNoiseModes {
SAMPLE_NONE,
SAMPLE_START,
SAMPLE_STOP,
NB_SAMPLEMODES
};
enum OutModes {
IN_MODE,
OUT_MODE,
NOISE_MODE,
NB_MODES
};
enum NoiseLinkType {
NONE_LINK,
MIN_LINK,
MAX_LINK,
AVERAGE_LINK,
NB_LINK
};
enum NoiseType {
WHITE_NOISE,
VINYL_NOISE,
SHELLAC_NOISE,
CUSTOM_NOISE,
NB_NOISE
};
typedef struct DeNoiseChannel {
double band_noise[NB_PROFILE_BANDS];
double noise_band_auto_var[NB_PROFILE_BANDS];
double noise_band_sample[NB_PROFILE_BANDS];
double *amt;
double *band_amt;
double *band_excit;
double *gain;
double *smoothed_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;
void *fft_in;
void *fft_out;
AVTXContext *fft, *ifft;
av_tx_fn tx_fn, itx_fn;
double noise_band_norm[NB_PROFILE_BANDS];
double noise_band_avr[NB_PROFILE_BANDS];
double noise_band_avi[NB_PROFILE_BANDS];
double noise_band_var[NB_PROFILE_BANDS];
double noise_reduction;
double last_noise_reduction;
double noise_floor;
double last_noise_floor;
double residual_floor;
double last_residual_floor;
double max_gain;
double max_var;
double gain_scale;
} DeNoiseChannel;
typedef struct AudioFFTDeNoiseContext {
const AVClass *class;
int format;
size_t sample_size;
size_t complex_sample_size;
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;
int noise_floor_link;
float ratio;
int gain_smooth;
float band_multiplier;
float floor_offset;
int channels;
int sample_noise;
int sample_noise_blocks;
int sample_noise_mode;
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[NB_PROFILE_BANDS];
int *bin2band;
double *window;
double *band_alpha;
double *band_beta;
DeNoiseChannel *dnch;
AVFrame *winframe;
double window_weight;
double floor;
double sample_floor;
int noise_band_edge[NB_PROFILE_BANDS + 2];
int noise_band_count;
double matrix_a[SOLVE_SIZE * SOLVE_SIZE];
double vector_b[SOLVE_SIZE];
double matrix_b[SOLVE_SIZE * NB_PROFILE_BANDS];
double matrix_c[SOLVE_SIZE * NB_PROFILE_BANDS];
} 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[] = {
{ "noise_reduction", "set the noise reduction",OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT,{.dbl = 12}, .01, 97, AFR },
{ "nr", "set the noise reduction", OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT, {.dbl = 12}, .01, 97, AFR },
{ "noise_floor", "set the noise floor",OFFSET(noise_floor), AV_OPT_TYPE_FLOAT, {.dbl =-50}, -80,-20, AFR },
{ "nf", "set the noise floor", OFFSET(noise_floor), AV_OPT_TYPE_FLOAT, {.dbl =-50}, -80,-20, AFR },
{ "noise_type", "set the noise type", OFFSET(noise_type), AV_OPT_TYPE_INT, {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, .unit = "type" },
{ "nt", "set the noise type", OFFSET(noise_type), AV_OPT_TYPE_INT, {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, .unit = "type" },
{ "white", "white noise", 0, AV_OPT_TYPE_CONST, {.i64 = WHITE_NOISE}, 0, 0, AF, .unit = "type" },
{ "w", "white noise", 0, AV_OPT_TYPE_CONST, {.i64 = WHITE_NOISE}, 0, 0, AF, .unit = "type" },
{ "vinyl", "vinyl noise", 0, AV_OPT_TYPE_CONST, {.i64 = VINYL_NOISE}, 0, 0, AF, .unit = "type" },
{ "v", "vinyl noise", 0, AV_OPT_TYPE_CONST, {.i64 = VINYL_NOISE}, 0, 0, AF, .unit = "type" },
{ "shellac", "shellac noise", 0, AV_OPT_TYPE_CONST, {.i64 = SHELLAC_NOISE}, 0, 0, AF, .unit = "type" },
{ "s", "shellac noise", 0, AV_OPT_TYPE_CONST, {.i64 = SHELLAC_NOISE}, 0, 0, AF, .unit = "type" },
{ "custom", "custom noise", 0, AV_OPT_TYPE_CONST, {.i64 = CUSTOM_NOISE}, 0, 0, AF, .unit = "type" },
{ "c", "custom noise", 0, AV_OPT_TYPE_CONST, {.i64 = CUSTOM_NOISE}, 0, 0, AF, .unit = "type" },
{ "band_noise", "set the custom bands noise", OFFSET(band_noise_str), AV_OPT_TYPE_STRING, {.str = 0}, 0, 0, AF },
{ "bn", "set the custom bands noise", OFFSET(band_noise_str), AV_OPT_TYPE_STRING, {.str = 0}, 0, 0, AF },
{ "residual_floor", "set the residual floor",OFFSET(residual_floor), AV_OPT_TYPE_FLOAT, {.dbl =-38}, -80,-20, AFR },
{ "rf", "set the residual floor", OFFSET(residual_floor), AV_OPT_TYPE_FLOAT, {.dbl =-38}, -80,-20, AFR },
{ "track_noise", "track noise", OFFSET(track_noise), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
{ "tn", "track noise", OFFSET(track_noise), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
{ "track_residual", "track residual", OFFSET(track_residual), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
{ "tr", "track residual", OFFSET(track_residual), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
{ "output_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64 = OUT_MODE}, 0, NB_MODES-1, AFR, .unit = "mode" },
{ "om", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64 = OUT_MODE}, 0, NB_MODES-1, AFR, .unit = "mode" },
{ "input", "input", 0, AV_OPT_TYPE_CONST, {.i64 = IN_MODE}, 0, 0, AFR, .unit = "mode" },
{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64 = IN_MODE}, 0, 0, AFR, .unit = "mode" },
{ "output", "output", 0, AV_OPT_TYPE_CONST, {.i64 = OUT_MODE}, 0, 0, AFR, .unit = "mode" },
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64 = OUT_MODE}, 0, 0, AFR, .unit = "mode" },
{ "noise", "noise", 0, AV_OPT_TYPE_CONST, {.i64 = NOISE_MODE}, 0, 0, AFR, .unit = "mode" },
{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64 = NOISE_MODE}, 0, 0, AFR, .unit = "mode" },
{ "adaptivity", "set adaptivity factor",OFFSET(ratio), AV_OPT_TYPE_FLOAT, {.dbl = 0.5}, 0, 1, AFR },
{ "ad", "set adaptivity factor",OFFSET(ratio), AV_OPT_TYPE_FLOAT, {.dbl = 0.5}, 0, 1, AFR },
{ "floor_offset", "set noise floor offset factor",OFFSET(floor_offset), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, -2, 2, AFR },
{ "fo", "set noise floor offset factor",OFFSET(floor_offset), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, -2, 2, AFR },
{ "noise_link", "set the noise floor link",OFFSET(noise_floor_link),AV_OPT_TYPE_INT,{.i64 = MIN_LINK}, 0, NB_LINK-1, AFR, .unit = "link" },
{ "nl", "set the noise floor link", OFFSET(noise_floor_link),AV_OPT_TYPE_INT,{.i64 = MIN_LINK}, 0, NB_LINK-1, AFR, .unit = "link" },
{ "none", "none", 0, AV_OPT_TYPE_CONST, {.i64 = NONE_LINK}, 0, 0, AFR, .unit = "link" },
{ "min", "min", 0, AV_OPT_TYPE_CONST, {.i64 = MIN_LINK}, 0, 0, AFR, .unit = "link" },
{ "max", "max", 0, AV_OPT_TYPE_CONST, {.i64 = MAX_LINK}, 0, 0, AFR, .unit = "link" },
{ "average", "average", 0, AV_OPT_TYPE_CONST, {.i64 = AVERAGE_LINK}, 0, 0, AFR, .unit = "link" },
{ "band_multiplier", "set band multiplier",OFFSET(band_multiplier), AV_OPT_TYPE_FLOAT,{.dbl = 1.25}, 0.2,5, AF },
{ "bm", "set band multiplier", OFFSET(band_multiplier), AV_OPT_TYPE_FLOAT,{.dbl = 1.25}, 0.2,5, AF },
{ "sample_noise", "set sample noise mode",OFFSET(sample_noise_mode),AV_OPT_TYPE_INT,{.i64 = SAMPLE_NONE}, 0, NB_SAMPLEMODES-1, AFR, .unit = "sample" },
{ "sn", "set sample noise mode",OFFSET(sample_noise_mode),AV_OPT_TYPE_INT,{.i64 = SAMPLE_NONE}, 0, NB_SAMPLEMODES-1, AFR, .unit = "sample" },
{ "none", "none", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_NONE}, 0, 0, AFR, .unit = "sample" },
{ "start", "start", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_START}, 0, 0, AFR, .unit = "sample" },
{ "begin", "start", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_START}, 0, 0, AFR, .unit = "sample" },
{ "stop", "stop", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_STOP}, 0, 0, AFR, .unit = "sample" },
{ "end", "stop", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_STOP}, 0, 0, AFR, .unit = "sample" },
{ "gain_smooth", "set gain smooth radius",OFFSET(gain_smooth), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 50, AFR },
{ "gs", "set gain smooth radius",OFFSET(gain_smooth), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 50, AFR },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afftdn);
static double 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 -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 double process_get_band_noise(AudioFFTDeNoiseContext *s,
DeNoiseChannel *dnch,
int band)
{
double product, sum, f;
int i = 0;
if (band < NB_PROFILE_BANDS)
return dnch->band_noise[band];
for (int j = 0; j < SOLVE_SIZE; j++) {
sum = 0.0;
for (int k = 0; k < NB_PROFILE_BANDS; k++)
sum += s->matrix_b[i++] * dnch->band_noise[k];
s->vector_b[j] = sum;
}
solve(s->matrix_a, s->vector_b, SOLVE_SIZE);
f = (0.5 * s->sample_rate) / s->band_centre[NB_PROFILE_BANDS-1];
f = 15.0 + log(f / 1.5) / log(1.5);
sum = 0.0;
product = 1.0;
for (int j = 0; j < SOLVE_SIZE; j++) {
sum += product * s->vector_b[j];
product *= f;
}
return sum;
}
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 spectral_flatness(AudioFFTDeNoiseContext *s, const double *const spectral,
double floor, int len, double *rnum, double *rden)
{
double num = 0., den = 0.;
int size = 0;
for (int n = 0; n < len; n++) {
const double v = spectral[n];
if (v > floor) {
num += log(v);
den += v;
size++;
}
}
size = FFMAX(size, 1);
num /= size;
den /= size;
num = exp(num);
*rnum = num;
*rden = den;
}
static void set_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int update_var, int update_auto_var);
static double floor_offset(const double *S, int size, double mean)
{
double offset = 0.0;
for (int n = 0; n < size; n++) {
const double p = S[n] - mean;
offset = fmax(offset, fabs(p));
}
return offset / mean;
}
static void process_frame(AVFilterContext *ctx,
AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch,
double *prior, double *prior_band_excit, int track_noise)
{
AVFilterLink *outlink = ctx->outputs[0];
const double *abs_var = dnch->abs_var;
const double ratio = outlink->frame_count_out ? s->ratio : 1.0;
const double rratio = 1. - ratio;
const int *bin2band = s->bin2band;
double *noisy_data = dnch->noisy_data;
double *band_excit = dnch->band_excit;
double *band_amt = dnch->band_amt;
double *smoothed_gain = dnch->smoothed_gain;
AVComplexDouble *fft_data_dbl = dnch->fft_out;
AVComplexFloat *fft_data_flt = dnch->fft_out;
double *gain = dnch->gain;
for (int i = 0; i < s->bin_count; i++) {
double sqr_new_gain, new_gain, power, mag, mag_abs_var, new_mag_abs_var;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
noisy_data[i] = mag = hypot(fft_data_flt[i].re, fft_data_flt[i].im);
break;
case AV_SAMPLE_FMT_DBLP:
noisy_data[i] = mag = hypot(fft_data_dbl[i].re, fft_data_dbl[i].im);
break;
}
power = mag * mag;
mag_abs_var = power / abs_var[i];
new_mag_abs_var = ratio * prior[i] + rratio * fmax(mag_abs_var - 1.0, 0.0);
new_gain = new_mag_abs_var / (1.0 + new_mag_abs_var);
sqr_new_gain = new_gain * new_gain;
prior[i] = mag_abs_var * sqr_new_gain;
dnch->clean_data[i] = power * sqr_new_gain;
gain[i] = new_gain;
}
if (track_noise) {
double flatness, num, den;
spectral_flatness(s, noisy_data, s->floor, s->bin_count, &num, &den);
flatness = num / den;
if (flatness > 0.8) {
const double offset = s->floor_offset * floor_offset(noisy_data, s->bin_count, den);
const double new_floor = av_clipd(10.0 * log10(den) - 100.0 + offset, -90., -20.);
dnch->noise_floor = 0.1 * new_floor + dnch->noise_floor * 0.9;
set_parameters(s, dnch, 1, 1);
}
}
for (int i = 0; i < s->number_of_bands; i++) {
band_excit[i] = 0.0;
band_amt[i] = 0.0;
}
for (int i = 0; i < s->bin_count; i++)
band_excit[bin2band[i]] += dnch->clean_data[i];
for (int i = 0; i < s->number_of_bands; i++) {
band_excit[i] = fmax(band_excit[i],
s->band_alpha[i] * band_excit[i] +
s->band_beta[i] * prior_band_excit[i]);
prior_band_excit[i] = 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++) {
band_amt[j] += dnch->spread_function[i++] * band_excit[k];
}
}
for (int i = 0; i < s->bin_count; i++)
dnch->amt[i] = band_amt[bin2band[i]];
for (int i = 0; i < s->bin_count; i++) {
if (dnch->amt[i] > abs_var[i]) {
gain[i] = 1.0;
} else if (dnch->amt[i] > dnch->min_abs_var[i]) {
const double limit = sqrt(abs_var[i] / dnch->amt[i]);
gain[i] = limit_gain(gain[i], limit);
} else {
gain[i] = limit_gain(gain[i], dnch->max_gain);
}
}
memcpy(smoothed_gain, gain, s->bin_count * sizeof(*smoothed_gain));
if (s->gain_smooth > 0) {
const int r = s->gain_smooth;
for (int i = r; i < s->bin_count - r; i++) {
const double gc = gain[i];
double num = 0., den = 0.;
for (int j = -r; j <= r; j++) {
const double g = gain[i + j];
const double d = 1. - fabs(g - gc);
num += g * d;
den += d;
}
smoothed_gain[i] = num / den;
}
}
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int i = 0; i < s->bin_count; i++) {
const float new_gain = smoothed_gain[i];
fft_data_flt[i].re *= new_gain;
fft_data_flt[i].im *= new_gain;
}
break;
case AV_SAMPLE_FMT_DBLP:
for (int i = 0; i < s->bin_count; i++) {
const double new_gain = smoothed_gain[i];
fft_data_dbl[i].re *= new_gain;
fft_data_dbl[i].im *= new_gain;
}
break;
}
}
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 == NB_PROFILE_BANDS) {
i = lrint(s->band_centre[NB_PROFILE_BANDS - 1] * 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->bin_count; m++) {
if (m == j) {
i = j;
d5 = band_noise;
if (k >= NB_PROFILE_BANDS) {
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);
}
for (i = 0; i < NB_PROFILE_BANDS; i++)
dnch->noise_band_auto_var[i] = dnch->max_var * exp((process_get_band_noise(s, dnch, i) - 2.0) * C);
}
static void read_custom_noise(AudioFFTDeNoiseContext *s, int ch)
{
DeNoiseChannel *dnch = &s->dnch[ch];
char *custom_noise_str, *p, *arg, *saveptr = NULL;
double band_noise[NB_PROFILE_BANDS] = { 0.f };
int ret;
if (!s->band_noise_str)
return;
custom_noise_str = p = av_strdup(s->band_noise_str);
if (!p)
return;
for (int i = 0; i < NB_PROFILE_BANDS; i++) {
float noise;
if (!(arg = av_strtok(p, "| ", &saveptr)))
break;
p = NULL;
ret = av_sscanf(arg, "%f", &noise);
if (ret != 1) {
av_log(s, AV_LOG_ERROR, "Custom band noise must be float.\n");
break;
}
band_noise[i] = av_clipd(noise, -24., 24.);
}
av_free(custom_noise_str);
memcpy(dnch->band_noise, band_noise, sizeof(band_noise));
}
static void set_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int update_var, int update_auto_var)
{
if (dnch->last_noise_floor != dnch->noise_floor)
dnch->last_noise_floor = dnch->noise_floor;
if (s->track_residual)
dnch->last_noise_floor = fmax(dnch->last_noise_floor, dnch->residual_floor);
dnch->max_var = s->floor * exp((100.0 + dnch->last_noise_floor) * C);
if (update_auto_var) {
for (int i = 0; i < NB_PROFILE_BANDS; i++)
dnch->noise_band_auto_var[i] = dnch->max_var * exp((process_get_band_noise(s, dnch, i) - 2.0) * C);
}
if (s->track_residual) {
if (update_var || dnch->last_residual_floor != dnch->residual_floor) {
update_var = 1;
dnch->last_residual_floor = dnch->residual_floor;
dnch->last_noise_reduction = fmax(dnch->last_noise_floor - dnch->last_residual_floor + 100., 0);
dnch->max_gain = exp(dnch->last_noise_reduction * (0.5 * C));
}
} else if (update_var || dnch->noise_reduction != dnch->last_noise_reduction) {
update_var = 1;
dnch->last_noise_reduction = dnch->noise_reduction;
dnch->last_residual_floor = av_clipd(dnch->last_noise_floor - dnch->last_noise_reduction, -80, -20);
dnch->max_gain = exp(dnch->last_noise_reduction * (0.5 * C));
}
dnch->gain_scale = 1.0 / (dnch->max_gain * dnch->max_gain);
if (update_var) {
set_band_parameters(s, dnch);
for (int i = 0; i < s->bin_count; i++) {
dnch->abs_var[i] = fmax(dnch->max_var * dnch->rel_var[i], 1.0);
dnch->min_abs_var[i] = dnch->gain_scale * dnch->abs_var[i];
}
}
}
static void reduce_mean(double *band_noise)
{
double mean = 0.f;
for (int i = 0; i < NB_PROFILE_BANDS; i++)
mean += band_noise[i];
mean /= NB_PROFILE_BANDS;
for (int i = 0; i < NB_PROFILE_BANDS; i++)
band_noise[i] -= mean;
}
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, ret, tx_type;
double dscale = 1.;
float fscale = 1.f;
void *scale;
s->format = inlink->format;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
s->sample_size = sizeof(float);
s->complex_sample_size = sizeof(AVComplexFloat);
tx_type = AV_TX_FLOAT_RDFT;
scale = &fscale;
break;
case AV_SAMPLE_FMT_DBLP:
s->sample_size = sizeof(double);
s->complex_sample_size = sizeof(AVComplexDouble);
tx_type = AV_TX_DOUBLE_RDFT;
scale = &dscale;
break;
}
s->dnch = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->dnch));
if (!s->dnch)
return AVERROR(ENOMEM);
s->channels = inlink->ch_layout.nb_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;
s->buffer_length = s->fft_length * 2;
s->bin_count = s->fft_length2 / 2 + 1;
s->band_centre[0] = 80;
for (i = 1; i < NB_PROFILE_BANDS; 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 < SOLVE_SIZE; j++) {
for (k = 0; k < SOLVE_SIZE; k++) {
s->matrix_a[j + k * SOLVE_SIZE] = 0.0;
for (m = 0; m < NB_PROFILE_BANDS; m++)
s->matrix_a[j + k * SOLVE_SIZE] += pow(m, j + k);
}
}
factor(s->matrix_a, SOLVE_SIZE);
i = 0;
for (j = 0; j < SOLVE_SIZE; j++)
for (k = 0; k < NB_PROFILE_BANDS; k++)
s->matrix_b[i++] = pow(k, j);
i = 0;
for (j = 0; j < NB_PROFILE_BANDS; j++)
for (k = 0; k < SOLVE_SIZE; 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->band_multiplier;
for (i = 0; i < s->bin_count; i++)
s->bin2band[i] = lrint(sdiv * freq2bark((0.5 * i * s->sample_rate) / s->fft_length2));
s->number_of_bands = s->bin2band[s->bin_count - 1] + 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->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
switch (s->noise_type) {
case WHITE_NOISE:
for (i = 0; i < NB_PROFILE_BANDS; i++)
dnch->band_noise[i] = 0.;
break;
case VINYL_NOISE:
for (i = 0; i < NB_PROFILE_BANDS; i++)
dnch->band_noise[i] = get_band_noise(s, i, 50.0, 500.5, 2125.0);
break;
case SHELLAC_NOISE:
for (i = 0; i < NB_PROFILE_BANDS; i++)
dnch->band_noise[i] = get_band_noise(s, i, 1.0, 500.0, 1.0E10);
break;
case CUSTOM_NOISE:
read_custom_noise(s, ch);
break;
default:
return AVERROR_BUG;
}
reduce_mean(dnch->band_noise);
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->smoothed_gain = av_calloc(s->bin_count, sizeof(*dnch->smoothed_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_in = av_calloc(s->fft_length2, s->sample_size);
dnch->fft_out = av_calloc(s->fft_length2 + 1, s->complex_sample_size);
ret = av_tx_init(&dnch->fft, &dnch->tx_fn, tx_type, 0, s->fft_length2, scale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&dnch->ifft, &dnch->itx_fn, tx_type, 1, s->fft_length2, scale, 0);
if (ret < 0)
return ret;
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->smoothed_gain ||
!dnch->prior ||
!dnch->prior_band_excit ||
!dnch->clean_data ||
!dnch->noisy_data ||
!dnch->out_samples ||
!dnch->fft_in ||
!dnch->fft_out ||
!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->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
double *prior_band_excit = dnch->prior_band_excit;
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->bin_count; 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->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->bin_count; 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];
}
}
s->winframe = ff_get_audio_buffer(inlink, s->window_length);
if (!s->winframe)
return AVERROR(ENOMEM);
wscale = sqrt(8.0 / (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);
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
dnch->noise_reduction = s->noise_reduction;
dnch->noise_floor = s->noise_floor;
dnch->residual_floor = s->residual_floor;
set_parameters(s, dnch, 1, 1);
}
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 < NB_PROFILE_BANDS + 1; 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[NB_PROFILE_BANDS + 1] = i;
}
s->noise_band_count = s->noise_band_edge[NB_PROFILE_BANDS + 1];
return 0;
}
static void init_sample_noise(DeNoiseChannel *dnch)
{
for (int i = 0; i < NB_PROFILE_BANDS; 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)
{
double *src_dbl = (double *)in->extended_data[ch];
float *src_flt = (float *)in->extended_data[ch];
double mag2, var = 0.0, avr = 0.0, avi = 0.0;
AVComplexDouble *fft_out_dbl = dnch->fft_out;
AVComplexFloat *fft_out_flt = dnch->fft_out;
double *fft_in_dbl = dnch->fft_in;
float *fft_in_flt = dnch->fft_in;
int edge, j, k, n, edgemax;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int i = 0; i < s->window_length; i++)
fft_in_flt[i] = s->window[i] * src_flt[i] * (1LL << 23);
for (int i = s->window_length; i < s->fft_length2; i++)
fft_in_flt[i] = 0.f;
break;
case AV_SAMPLE_FMT_DBLP:
for (int i = 0; i < s->window_length; i++)
fft_in_dbl[i] = s->window[i] * src_dbl[i] * (1LL << 23);
for (int i = s->window_length; i < s->fft_length2; i++)
fft_in_dbl[i] = 0.;
break;
}
dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, s->sample_size);
edge = s->noise_band_edge[0];
j = edge;
k = 0;
n = j;
edgemax = fmin(s->fft_length2, s->noise_band_edge[NB_PROFILE_BANDS]);
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 == NB_PROFILE_BANDS) {
j++;
}
var = 0.0;
avr = 0.0;
avi = 0.0;
}
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
avr += fft_out_flt[n].re;
avi += fft_out_flt[n].im;
mag2 = fft_out_flt[n].re * fft_out_flt[n].re +
fft_out_flt[n].im * fft_out_flt[n].im;
break;
case AV_SAMPLE_FMT_DBLP:
avr += fft_out_dbl[n].re;
avi += fft_out_dbl[n].im;
mag2 = fft_out_dbl[n].re * fft_out_dbl[n].re +
fft_out_dbl[n].im * fft_out_dbl[n].im;
break;
}
mag2 = fmax(mag2, s->sample_floor);
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] = 10.0 * log10(dnch->noise_band_var[i] / s->floor) - 100.0;
}
if (s->noise_band_count < NB_PROFILE_BANDS) {
for (int i = s->noise_band_count; i < NB_PROFILE_BANDS; i++)
sample_noise[i] = sample_noise[i - 1];
}
}
static void set_noise_profile(AudioFFTDeNoiseContext *s,
DeNoiseChannel *dnch,
double *sample_noise)
{
double new_band_noise[NB_PROFILE_BANDS];
double temp[NB_PROFILE_BANDS];
double sum = 0.0;
for (int m = 0; m < NB_PROFILE_BANDS; m++)
temp[m] = sample_noise[m];
for (int m = 0, i = 0; m < SOLVE_SIZE; m++) {
sum = 0.0;
for (int n = 0; n < NB_PROFILE_BANDS; n++)
sum += s->matrix_b[i++] * temp[n];
s->vector_b[m] = sum;
}
solve(s->matrix_a, s->vector_b, SOLVE_SIZE);
for (int m = 0, i = 0; m < NB_PROFILE_BANDS; m++) {
sum = 0.0;
for (int n = 0; n < SOLVE_SIZE; n++)
sum += s->matrix_c[i++] * s->vector_b[n];
temp[m] = sum;
}
reduce_mean(temp);
av_log(s, AV_LOG_INFO, "bn=");
for (int m = 0; m < NB_PROFILE_BANDS; m++) {
new_band_noise[m] = temp[m];
new_band_noise[m] = av_clipd(new_band_noise[m], -24.0, 24.0);
av_log(s, AV_LOG_INFO, "%f ", new_band_noise[m]);
}
av_log(s, AV_LOG_INFO, "\n");
memcpy(dnch->band_noise, new_band_noise, sizeof(new_band_noise));
}
static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioFFTDeNoiseContext *s = ctx->priv;
AVFrame *in = arg;
const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
const int window_length = s->window_length;
const double *window = s->window;
for (int ch = start; ch < end; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
const double *src_dbl = (const double *)in->extended_data[ch];
const float *src_flt = (const float *)in->extended_data[ch];
double *dst = dnch->out_samples;
double *fft_in_dbl = dnch->fft_in;
float *fft_in_flt = dnch->fft_in;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < window_length; m++)
fft_in_flt[m] = window[m] * src_flt[m] * (1LL << 23);
for (int m = window_length; m < s->fft_length2; m++)
fft_in_flt[m] = 0.f;
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < window_length; m++)
fft_in_dbl[m] = window[m] * src_dbl[m] * (1LL << 23);
for (int m = window_length; m < s->fft_length2; m++)
fft_in_dbl[m] = 0.;
break;
}
dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, s->sample_size);
process_frame(ctx, s, dnch,
dnch->prior,
dnch->prior_band_excit,
s->track_noise);
dnch->itx_fn(dnch->ifft, dnch->fft_in, dnch->fft_out, s->complex_sample_size);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < window_length; m++)
dst[m] += s->window[m] * fft_in_flt[m] / (1LL << 23);
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < window_length; m++)
dst[m] += s->window[m] * fft_in_dbl[m] / (1LL << 23);
break;
}
}
return 0;
}
static int output_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioFFTDeNoiseContext *s = ctx->priv;
const int output_mode = ctx->is_disabled ? IN_MODE : s->output_mode;
const int offset = s->window_length - s->sample_advance;
AVFrame *out;
for (int ch = 0; ch < s->channels; ch++) {
uint8_t *src = (uint8_t *)s->winframe->extended_data[ch];
memmove(src, src + s->sample_advance * s->sample_size,
offset * s->sample_size);
memcpy(src + offset * s->sample_size, in->extended_data[ch],
in->nb_samples * s->sample_size);
memset(src + s->sample_size * (offset + in->nb_samples), 0,
(s->sample_advance - in->nb_samples) * s->sample_size);
}
if (s->track_noise) {
double average = 0.0, min = DBL_MAX, max = -DBL_MAX;
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
average += dnch->noise_floor;
max = fmax(max, dnch->noise_floor);
min = fmin(min, dnch->noise_floor);
}
average /= inlink->ch_layout.nb_channels;
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
switch (s->noise_floor_link) {
case MIN_LINK: dnch->noise_floor = min; break;
case MAX_LINK: dnch->noise_floor = max; break;
case AVERAGE_LINK: dnch->noise_floor = average; break;
case NONE_LINK:
default:
break;
}
if (dnch->noise_floor != dnch->last_noise_floor)
set_parameters(s, dnch, 1, 0);
}
}
if (s->sample_noise_mode == SAMPLE_START) {
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
init_sample_noise(dnch);
}
s->sample_noise_mode = SAMPLE_NONE;
s->sample_noise = 1;
s->sample_noise_blocks = 0;
}
if (s->sample_noise) {
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
sample_noise_block(s, dnch, s->winframe, ch);
}
s->sample_noise_blocks++;
}
if (s->sample_noise_mode == SAMPLE_STOP) {
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
double sample_noise[NB_PROFILE_BANDS];
if (s->sample_noise_blocks <= 0)
break;
finish_sample_noise(s, dnch, sample_noise);
set_noise_profile(s, dnch, sample_noise);
set_parameters(s, dnch, 1, 1);
}
s->sample_noise = 0;
s->sample_noise_blocks = 0;
s->sample_noise_mode = SAMPLE_NONE;
}
ff_filter_execute(ctx, filter_channel, s->winframe, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
double *src = dnch->out_samples;
const double *orig_dbl = (const double *)s->winframe->extended_data[ch];
const float *orig_flt = (const float *)s->winframe->extended_data[ch];
double *dst_dbl = (double *)out->extended_data[ch];
float *dst_flt = (float *)out->extended_data[ch];
switch (output_mode) {
case IN_MODE:
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < out->nb_samples; m++)
dst_flt[m] = orig_flt[m];
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < out->nb_samples; m++)
dst_dbl[m] = orig_dbl[m];
break;
}
break;
case OUT_MODE:
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < out->nb_samples; m++)
dst_flt[m] = src[m];
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < out->nb_samples; m++)
dst_dbl[m] = src[m];
break;
}
break;
case NOISE_MODE:
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < out->nb_samples; m++)
dst_flt[m] = orig_flt[m] - src[m];
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < out->nb_samples; m++)
dst_dbl[m] = orig_dbl[m] - src[m];
break;
}
break;
default:
if (in != out)
av_frame_free(&in);
av_frame_free(&out);
return AVERROR_BUG;
}
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));
}
if (out != in)
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];
AudioFFTDeNoiseContext *s = ctx->priv;
AVFrame *in = NULL;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in);
if (ret < 0)
return ret;
if (ret > 0)
return output_frame(inlink, in);
if (ff_inlink_queued_samples(inlink) >= s->sample_advance) {
ff_filter_set_ready(ctx, 10);
return 0;
}
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
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);
av_frame_free(&s->winframe);
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->smoothed_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_in);
av_freep(&dnch->fft_out);
av_tx_uninit(&dnch->fft);
av_tx_uninit(&dnch->ifft);
}
av_freep(&s->dnch);
}
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
AudioFFTDeNoiseContext *s = ctx->priv;
int ret = 0;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
if (!strcmp(cmd, "sample_noise") || !strcmp(cmd, "sn"))
return 0;
for (int ch = 0; ch < s->channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
dnch->noise_reduction = s->noise_reduction;
dnch->noise_floor = s->noise_floor;
dnch->residual_floor = s->residual_floor;
set_parameters(s, dnch, 1, 1);
}
return 0;
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
};
const AVFilter ff_af_afftdn = {
.name = "afftdn",
.description = NULL_IF_CONFIG_SMALL("Denoise audio samples using FFT."),
.priv_size = sizeof(AudioFFTDeNoiseContext),
.priv_class = &afftdn_class,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
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