ffmpeg/FFmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-10-19 04:53:25 +00:00
Code Issues Packages Projects Releases Wiki Activity
20e3316584
FFmpeg/libavfilter/af_afftdn.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback 
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +02:00

1426 lines
44 KiB
C
Raw Blame History

/*
* 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 "libavutil/tx.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;
AVComplexFloat *fft_in;
AVComplexFloat *fft_out;
AVTXContext *fft, *ifft;
av_tx_fn tx_fn, itx_fn;
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,
AVComplexFloat *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, ret;
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];
float scale;
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_in = av_calloc(s->fft_length2 + 1, sizeof(*dnch->fft_in));
dnch->fft_out = av_calloc(s->fft_length2 + 1, sizeof(*dnch->fft_out));
ret = av_tx_init(&dnch->fft, &dnch->tx_fn, AV_TX_FLOAT_FFT, 0, s->fft_length2, &scale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&dnch->ifft, &dnch->itx_fn, AV_TX_FLOAT_FFT, 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->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->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(AVComplexFloat *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(AVComplexFloat *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_in[i].re = s->window[i] * src[i] * (1LL << 24);
dnch->fft_in[i].im = 0.0;
}
for (int i = s->window_length; i < s->fft_length2; i++) {
dnch->fft_in[i].re = 0.0;
dnch->fft_in[i].im = 0.0;
}
dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, sizeof(float));
preprocess(dnch->fft_out, 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_out[s->fft_length2].re = dnch->fft_out[0].im;
dnch->fft_out[0].im = 0.0;
dnch->fft_out[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_out[n].re;
avi += dnch->fft_out[n].im;
mag2 = dnch->fft_out[n].re * dnch->fft_out[n].re +
dnch->fft_out[n].im * dnch->fft_out[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 = 0, n;
for (int m = 0; m < 15; m++)
temp[m] = sample_noise[m];
if (new_profile) {
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_in[m].re = s->window[m] * src[m] * (1LL << 24);
dnch->fft_in[m].im = 0;
}
for (int m = s->window_length; m < s->fft_length2; m++) {
dnch->fft_in[m].re = 0;
dnch->fft_in[m].im = 0;
}
dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, sizeof(float));
preprocess(dnch->fft_out, s->fft_length);
process_frame(s, dnch, dnch->fft_out,
dnch->prior,
dnch->prior_band_excit,
s->track_noise);
postprocess(dnch->fft_out, s->fft_length);
dnch->itx_fn(dnch->ifft, dnch->fft_in, dnch->fft_out, sizeof(float));
for (int m = 0; m < s->window_length; m++)
dst[m] += s->window[m] * dnch->fft_in[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;
const int output_mode = ctx->is_disabled ? IN_MODE : s->output_mode;
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;
ff_filter_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 (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_in);
av_freep(&dnch->fft_out);
av_tx_uninit(&dnch->fft);
av_tx_uninit(&dnch->ifft);
}
av_freep(&s->dnch);
}
av_audio_fifo_free(s->fifo);
}
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
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,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
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(outputs),
FILTER_QUERY_FUNC(query_formats),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
Reference in New Issue View Git Blame Copy Permalink