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FFmpeg/libavfilter/af_afftdn.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

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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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