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32 #define C (M_LN10 * 0.1)
33 #define SOLVE_SIZE (5)
34 #define NB_PROFILE_BANDS (15)
162 #define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x)
163 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
164 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
229 d1 =
a /
s->band_centre[band];
230 d1 = 10.0 * log(1.0 + d1 * d1) /
M_LN10;
231 d2 =
b /
s->band_centre[band];
232 d2 = 10.0 * log(1.0 + d2 * d2) /
M_LN10;
233 d3 =
s->band_centre[band] /
c;
234 d3 = 10.0 * log(1.0 + d3 * d3) /
M_LN10;
236 return -d1 + d2 - d3;
241 for (
int i = 0;
i <
size - 1;
i++) {
242 for (
int j =
i + 1; j <
size; j++) {
246 for (
int k =
i + 1; k <
size; k++) {
255 for (
int i = 0;
i <
size - 1;
i++) {
256 for (
int j =
i + 1; j <
size; j++) {
258 vector[j] -=
d * vector[
i];
264 for (
int i =
size - 2;
i >= 0;
i--) {
265 double d = vector[
i];
266 for (
int j =
i + 1; j <
size; j++)
276 double product, sum,
f;
286 s->vector_b[j] = sum;
291 f = 15.0 + log(
f / 1.5) / log(1.5);
295 sum += product *
s->vector_b[j];
305 return (
b *
a - 1.0) / (
b +
a - 2.0);
307 return (
b *
a - 2.0 *
a + 1.0) / (
b -
a);
312 double floor,
int len,
double *rnum,
double *rden)
314 double num = 0., den = 0.;
317 for (
int n = 0; n <
len; n++) {
318 const double v = spectral[n];
343 for (
int n = 0; n <
size; n++) {
344 const double p =
S[n] -
mean;
354 double *prior,
double *prior_band_excit,
int track_noise)
357 const double *abs_var = dnch->
abs_var;
359 const double rratio = 1. - ratio;
360 const int *bin2band =
s->bin2band;
367 double *gain = dnch->
gain;
369 for (
int i = 0;
i <
s->bin_count;
i++) {
370 double sqr_new_gain, new_gain,
power, mag, mag_abs_var, new_mag_abs_var;
374 noisy_data[
i] = mag =
hypot(fft_data_flt[
i].
re, fft_data_flt[
i].
im);
377 noisy_data[
i] = mag =
hypot(fft_data_dbl[
i].
re, fft_data_dbl[
i].
im);
382 mag_abs_var =
power / abs_var[
i];
383 new_mag_abs_var = ratio * prior[
i] + rratio *
fmax(mag_abs_var - 1.0, 0.0);
384 new_gain = new_mag_abs_var / (1.0 + new_mag_abs_var);
385 sqr_new_gain = new_gain * new_gain;
386 prior[
i] = mag_abs_var * sqr_new_gain;
392 double flatness, num, den;
396 flatness = num / den;
397 if (flatness > 0.8) {
399 const double new_floor =
av_clipd(10.0 * log10(den) - 100.0 +
offset, -90., -20.);
406 for (
int i = 0;
i <
s->number_of_bands;
i++) {
411 for (
int i = 0;
i <
s->bin_count;
i++)
414 for (
int i = 0;
i <
s->number_of_bands;
i++) {
415 band_excit[
i] =
fmax(band_excit[
i],
416 s->band_alpha[
i] * band_excit[
i] +
417 s->band_beta[
i] * prior_band_excit[
i]);
418 prior_band_excit[
i] = band_excit[
i];
421 for (
int j = 0,
i = 0; j <
s->number_of_bands; j++) {
422 for (
int k = 0; k <
s->number_of_bands; k++) {
427 for (
int i = 0;
i <
s->bin_count;
i++)
428 dnch->
amt[
i] = band_amt[bin2band[
i]];
430 for (
int i = 0;
i <
s->bin_count;
i++) {
431 if (dnch->
amt[
i] > abs_var[
i]) {
434 const double limit = sqrt(abs_var[
i] / dnch->
amt[
i]);
442 memcpy(smoothed_gain, gain,
s->bin_count *
sizeof(*smoothed_gain));
443 if (
s->gain_smooth > 0) {
444 const int r =
s->gain_smooth;
446 for (
int i =
r;
i <
s->bin_count -
r;
i++) {
447 const double gc = gain[
i];
448 double num = 0., den = 0.;
450 for (
int j = -
r; j <=
r; j++) {
451 const double g = gain[
i + j];
452 const double d = 1. -
fabs(
g - gc);
458 smoothed_gain[
i] = num / den;
464 for (
int i = 0;
i <
s->bin_count;
i++) {
465 const float new_gain = smoothed_gain[
i];
467 fft_data_flt[
i].
re *= new_gain;
468 fft_data_flt[
i].
im *= new_gain;
472 for (
int i = 0;
i <
s->bin_count;
i++) {
473 const double new_gain = smoothed_gain[
i];
475 fft_data_dbl[
i].
re *= new_gain;
476 fft_data_dbl[
i].
im *= new_gain;
484 double d = x / 7500.0;
486 return 13.0 * atan(7.6
E-4 * x) + 3.5 * atan(
d *
d);
492 return lrint(
s->band_centre[0] / 1.5);
494 return s->band_centre[band];
504 i =
lrint(
s->band_centre[band] / 1.224745);
507 return FFMIN(
i,
s->sample_rate / 2);
513 double band_noise, d2, d3, d4, d5;
514 int i = 0, j = 0, k = 0;
518 for (
int m = j; m <
s->bin_count; m++) {
533 dnch->
rel_var[m] =
exp((d5 * d3 + band_noise * d4) *
C);
543 char *custom_noise_str, *p, *
arg, *saveptr =
NULL;
547 if (!
s->band_noise_str)
550 custom_noise_str = p =
av_strdup(
s->band_noise_str);
572 memcpy(dnch->
band_noise, band_noise,
sizeof(band_noise));
580 if (
s->track_residual)
584 if (update_auto_var) {
589 if (
s->track_residual) {
608 for (
int i = 0;
i <
s->bin_count;
i++) {
620 mean += band_noise[
i];
624 band_noise[
i] -=
mean;
631 double wscale, sar, sum, sdiv;
632 int i, j, k, m, n,
ret, tx_type;
641 s->sample_size =
sizeof(
float);
647 s->sample_size =
sizeof(
double);
658 s->channels =
inlink->ch_layout.nb_channels;
659 s->sample_rate =
inlink->sample_rate;
660 s->sample_advance =
s->sample_rate / 80;
661 s->window_length = 3 *
s->sample_advance;
662 s->fft_length2 = 1 << (32 -
ff_clz(
s->window_length));
663 s->fft_length =
s->fft_length2;
664 s->buffer_length =
s->fft_length * 2;
665 s->bin_count =
s->fft_length2 / 2 + 1;
667 s->band_centre[0] = 80;
669 s->band_centre[
i] =
lrint(1.5 *
s->band_centre[
i - 1] + 5.0);
670 if (
s->band_centre[
i] < 1000) {
671 s->band_centre[
i] = 10 * (
s->band_centre[
i] / 10);
672 }
else if (
s->band_centre[
i] < 5000) {
673 s->band_centre[
i] = 50 * ((
s->band_centre[
i] + 20) / 50);
674 }
else if (
s->band_centre[
i] < 15000) {
675 s->band_centre[
i] = 100 * ((
s->band_centre[
i] + 45) / 100);
677 s->band_centre[
i] = 1000 * ((
s->band_centre[
i] + 495) / 1000);
694 s->matrix_b[
i++] = pow(k, j);
699 s->matrix_c[
i++] = pow(j, k);
701 s->window =
av_calloc(
s->window_length,
sizeof(*
s->window));
702 s->bin2band =
av_calloc(
s->bin_count,
sizeof(*
s->bin2band));
703 if (!
s->window || !
s->bin2band)
706 sdiv =
s->band_multiplier;
707 for (
i = 0;
i <
s->bin_count;
i++)
710 s->number_of_bands =
s->bin2band[
s->bin_count - 1] + 1;
712 s->band_alpha =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_alpha));
713 s->band_beta =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_beta));
714 if (!
s->band_alpha || !
s->band_beta)
717 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
720 switch (
s->noise_type) {
787 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
793 p1 = pow(0.1, 2.5 / sdiv);
794 p2 = pow(0.1, 1.0 / sdiv);
796 for (m = 0; m <
s->number_of_bands; m++) {
797 for (n = 0; n <
s->number_of_bands; n++) {
808 for (m = 0; m <
s->number_of_bands; m++) {
810 prior_band_excit[m] = 0.0;
813 for (m = 0; m <
s->bin_count; m++)
817 for (m = 0; m <
s->number_of_bands; m++) {
818 for (n = 0; n <
s->number_of_bands; n++)
824 for (
int i = 0;
i <
s->number_of_bands;
i++) {
825 if (
i <
lrint(12.0 * sdiv)) {
828 dnch->
band_excit[
i] = pow(0.1, 2.5 - 0.2 * (
i / sdiv - 14.0));
833 for (
int i = 0;
i <
s->buffer_length;
i++)
837 for (
int i = 0;
i <
s->number_of_bands;
i++)
838 for (
int k = 0; k <
s->number_of_bands; k++)
843 sar =
s->sample_advance /
s->sample_rate;
844 for (
int i = 0;
i <
s->bin_count;
i++) {
845 if ((
i ==
s->fft_length2) || (
s->bin2band[
i] > j)) {
846 double d6 = (
i - 1) *
s->sample_rate /
s->fft_length;
847 double d7 =
fmin(0.008 + 2.2 / d6, 0.03);
848 s->band_alpha[j] =
exp(-sar / d7);
849 s->band_beta[j] = 1.0 -
s->band_alpha[j];
858 wscale = sqrt(8.0 / (9.0 *
s->fft_length));
860 for (
int i = 0;
i <
s->window_length;
i++) {
861 double d10 = sin(
i *
M_PI /
s->window_length);
867 s->window_weight = 0.5 * sum;
868 s->floor = (1LL << 48) *
exp(-23.025558369790467) *
s->window_weight;
869 s->sample_floor =
s->floor *
exp(4.144600506562284);
871 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
885 if (
s->noise_band_edge[j] >
lrint(1.1 *
s->noise_band_edge[j - 1]))
910 double mag2, var = 0.0, avr = 0.0, avi = 0.0;
913 double *fft_in_dbl = dnch->
fft_in;
914 float *fft_in_flt = dnch->
fft_in;
915 int edge, j, k, n, edgemax;
919 for (
int i = 0;
i <
s->window_length;
i++)
920 fft_in_flt[
i] =
s->window[
i] * src_flt[
i] * (1LL << 23);
922 for (
int i =
s->window_length; i < s->fft_length2;
i++)
926 for (
int i = 0;
i <
s->window_length;
i++)
927 fft_in_dbl[
i] =
s->window[
i] * src_dbl[
i] * (1LL << 23);
929 for (
int i =
s->window_length; i < s->fft_length2;
i++)
936 edge =
s->noise_band_edge[0];
941 for (
int i = j;
i <= edgemax;
i++) {
942 if ((
i == j) && (
i < edgemax)) {
951 j =
s->noise_band_edge[k];
962 avr += fft_out_flt[n].
re;
963 avi += fft_out_flt[n].
im;
964 mag2 = fft_out_flt[n].
re * fft_out_flt[n].
re +
965 fft_out_flt[n].
im * fft_out_flt[n].
im;
968 avr += fft_out_dbl[n].
re;
969 avi += fft_out_dbl[n].
im;
970 mag2 = fft_out_dbl[n].
re * fft_out_dbl[n].
re +
971 fft_out_dbl[n].
im * fft_out_dbl[n].
im;
975 mag2 =
fmax(mag2,
s->sample_floor);
989 double *sample_noise)
991 for (
int i = 0;
i <
s->noise_band_count;
i++) {
1002 sample_noise[
i] = sample_noise[
i - 1];
1008 double *sample_noise)
1015 temp[m] = sample_noise[m];
1020 sum +=
s->matrix_b[
i++] *
temp[n];
1021 s->vector_b[m] = sum;
1027 sum +=
s->matrix_c[
i++] *
s->vector_b[n];
1035 new_band_noise[m] =
temp[m];
1036 new_band_noise[m] =
av_clipd(new_band_noise[m], -24.0, 24.0);
1040 memcpy(dnch->
band_noise, new_band_noise,
sizeof(new_band_noise));
1049 const int window_length =
s->window_length;
1050 const double *
window =
s->window;
1052 for (
int ch = start; ch < end; ch++) {
1054 const double *src_dbl = (
const double *)in->
extended_data[ch];
1055 const float *src_flt = (
const float *)in->
extended_data[ch];
1057 double *fft_in_dbl = dnch->
fft_in;
1058 float *fft_in_flt = dnch->
fft_in;
1060 switch (
s->format) {
1062 for (
int m = 0; m < window_length; m++)
1063 fft_in_flt[m] =
window[m] * src_flt[m] * (1LL << 23);
1065 for (
int m = window_length; m <
s->fft_length2; m++)
1066 fft_in_flt[m] = 0.
f;
1069 for (
int m = 0; m < window_length; m++)
1070 fft_in_dbl[m] =
window[m] * src_dbl[m] * (1LL << 23);
1072 for (
int m = window_length; m <
s->fft_length2; m++)
1086 switch (
s->format) {
1088 for (
int m = 0; m < window_length; m++)
1089 dst[m] +=
s->window[m] * fft_in_flt[m] / (1LL << 23);
1092 for (
int m = 0; m < window_length; m++)
1093 dst[m] +=
s->window[m] * fft_in_dbl[m] / (1LL << 23);
1106 const int output_mode =
ctx->is_disabled ?
IN_MODE :
s->output_mode;
1107 const int offset =
s->window_length -
s->sample_advance;
1110 for (
int ch = 0; ch <
s->channels; ch++) {
1111 uint8_t *
src = (uint8_t *)
s->winframe->extended_data[ch];
1113 memmove(
src,
src +
s->sample_advance *
s->sample_size,
1118 (
s->sample_advance - in->
nb_samples) *
s->sample_size);
1121 if (
s->track_noise) {
1122 double average = 0.0,
min = DBL_MAX,
max = -DBL_MAX;
1124 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1132 average /=
inlink->ch_layout.nb_channels;
1134 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1137 switch (
s->noise_floor_link) {
1152 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1158 s->sample_noise = 1;
1159 s->sample_noise_blocks = 0;
1162 if (
s->sample_noise) {
1163 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1168 s->sample_noise_blocks++;
1172 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1176 if (
s->sample_noise_blocks <= 0)
1182 s->sample_noise = 0;
1183 s->sample_noise_blocks = 0;
1202 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1205 const double *orig_dbl = (
const double *)
s->winframe->extended_data[ch];
1206 const float *orig_flt = (
const float *)
s->winframe->extended_data[ch];
1207 double *dst_dbl = (
double *)
out->extended_data[ch];
1208 float *dst_flt = (
float *)
out->extended_data[ch];
1210 switch (output_mode) {
1212 switch (
s->format) {
1214 for (
int m = 0; m <
out->nb_samples; m++)
1215 dst_flt[m] = orig_flt[m];
1218 for (
int m = 0; m <
out->nb_samples; m++)
1219 dst_dbl[m] = orig_dbl[m];
1224 switch (
s->format) {
1226 for (
int m = 0; m <
out->nb_samples; m++)
1227 dst_flt[m] =
src[m];
1230 for (
int m = 0; m <
out->nb_samples; m++)
1231 dst_dbl[m] =
src[m];
1236 switch (
s->format) {
1238 for (
int m = 0; m <
out->nb_samples; m++)
1239 dst_flt[m] = orig_flt[m] -
src[m];
1242 for (
int m = 0; m <
out->nb_samples; m++)
1243 dst_dbl[m] = orig_dbl[m] -
src[m];
1254 memmove(
src,
src +
s->sample_advance, (
s->window_length -
s->sample_advance) *
sizeof(*
src));
1255 memset(
src + (
s->window_length -
s->sample_advance), 0,
s->sample_advance *
sizeof(*
src));
1301 for (
int ch = 0; ch <
s->channels; ch++) {
1327 char *res,
int res_len,
int flags)
1336 if (!strcmp(cmd,
"sample_noise") || !strcmp(cmd,
"sn"))
1339 for (
int ch = 0; ch <
s->channels; ch++) {
1371 .priv_class = &afftdn_class,
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
@ AV_SAMPLE_FMT_FLTP
float, planar
double noise_band_auto_var[NB_PROFILE_BANDS]
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static int activate(AVFilterContext *ctx)
static const AVFilterPad inputs[]
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
static void solve(double *matrix, double *vector, int size)
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
This structure describes decoded (raw) audio or video data.
static void process_frame(AVFilterContext *ctx, AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *prior, double *prior_band_excit, int track_noise)
static void sample_noise_block(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, AVFrame *in, int ch)
const char * name
Filter name.
int nb_channels
Number of channels in this layout.
A link between two filters.
double vector_b[SOLVE_SIZE]
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
double noise_band_norm[NB_PROFILE_BANDS]
static void factor(double *array, int size)
double noise_band_avr[NB_PROFILE_BANDS]
static int config_input(AVFilterLink *inlink)
static SDL_Window * window
size_t complex_sample_size
static double freq2bark(double x)
double band_noise[NB_PROFILE_BANDS]
AVChannelLayout ch_layout
Channel layout of the audio data.
static av_always_inline float scale(float x, float s)
A filter pad used for either input or output.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
double last_noise_reduction
static const AVOption afftdn_options[]
double matrix_a[SOLVE_SIZE *SOLVE_SIZE]
static __device__ float floor(float a)
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok().
double noise_band_sample[NB_PROFILE_BANDS]
#define FILTER_INPUTS(array)
static int get_band_edge(AudioFFTDeNoiseContext *s, int band)
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
Describe the class of an AVClass context structure.
static __device__ float fabs(float a)
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
double matrix_b[SOLVE_SIZE *NB_PROFILE_BANDS]
static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
static void set_band_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch)
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
static void init_sample_noise(DeNoiseChannel *dnch)
static void read_custom_noise(AudioFFTDeNoiseContext *s, int ch)
double last_residual_floor
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static const AVFilterPad outputs[]
static int output_frame(AVFilterLink *inlink, AVFrame *in)
double fmin(double, double)
static av_const double hypot(double x, double y)
double matrix_c[SOLVE_SIZE *NB_PROFILE_BANDS]
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
static void finish_sample_noise(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *sample_noise)
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
FF_FILTER_FORWARD_WANTED(outlink, inlink)
static double limit_gain(double a, double b)
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
#define AV_LOG_INFO
Standard information.
static av_cold void uninit(AVFilterContext *ctx)
int nb_samples
number of audio samples (per channel) described by this frame
#define i(width, name, range_min, range_max)
double noise_band_avi[NB_PROFILE_BANDS]
uint8_t ** extended_data
pointers to the data planes/channels.
double * prior_band_excit
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
const char * name
Pad name.
int ff_inlink_queued_samples(AVFilterLink *link)
void * av_calloc(size_t nmemb, size_t size)
static double limit(double x)
static int array[MAX_W *MAX_W]
static double get_band_noise(AudioFFTDeNoiseContext *s, int band, double a, double b, double c)
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
double fmax(double, double)
static float power(float r, float g, float b, float max)
static void set_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int update_var, int update_auto_var)
static double process_get_band_noise(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int band)
static int noise(AVBSFContext *ctx, AVPacket *pkt)
static void spectral_flatness(AudioFFTDeNoiseContext *s, const double *const spectral, double floor, int len, double *rnum, double *rden)
@ AV_SAMPLE_FMT_DBLP
double, planar
static float mean(const float *input, int size)
double noise_band_var[NB_PROFILE_BANDS]
int band_centre[NB_PROFILE_BANDS]
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
char * av_strdup(const char *s)
Duplicate a string.
AVFILTER_DEFINE_CLASS(afftdn)
AVChannelLayout ch_layout
channel layout of current buffer (see libavutil/channel_layout.h)
FF_FILTER_FORWARD_STATUS(inlink, outlink)
#define FILTER_OUTPUTS(array)
static double floor_offset(const double *S, int size, double mean)
int noise_band_edge[NB_PROFILE_BANDS+2]
static void reduce_mean(double *band_noise)
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
#define flags(name, subs,...)
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
static void set_noise_profile(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *sample_noise)
static av_always_inline int ff_filter_execute(AVFilterContext *ctx, avfilter_action_func *func, void *arg, int *ret, int nb_jobs)
static int get_band_centre(AudioFFTDeNoiseContext *s, int band)
const AVFilter ff_af_afftdn
#define FILTER_SAMPLEFMTS(...)
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.