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31 #define C (M_LN10 * 0.1)
32 #define SOLVE_SIZE (5)
33 #define NB_PROFILE_BANDS (15)
161 #define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x)
162 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
163 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
228 d1 =
a /
s->band_centre[band];
230 d2 =
b /
s->band_centre[band];
232 d3 =
s->band_centre[band] /
c;
235 return -d1 + d2 - d3;
240 for (
int i = 0;
i <
size - 1;
i++) {
241 for (
int j =
i + 1; j <
size; j++) {
245 for (
int k =
i + 1; k <
size; k++) {
254 for (
int i = 0;
i <
size - 1;
i++) {
255 for (
int j =
i + 1; j <
size; j++) {
257 vector[j] -=
d * vector[
i];
263 for (
int i =
size - 2;
i >= 0;
i--) {
264 double d = vector[
i];
265 for (
int j =
i + 1; j <
size; j++)
275 double product, sum,
f;
285 s->vector_b[j] = sum;
294 sum += product *
s->vector_b[j];
304 return (
b *
a - 1.0) / (
b +
a - 2.0);
306 return (
b *
a - 2.0 *
a + 1.0) / (
b -
a);
311 double floor,
int len,
double *rnum,
double *rden)
313 double num = 0., den = 0.;
316 for (
int n = 0; n <
len; n++) {
317 const double v = spectral[n];
342 for (
int n = 0; n <
size; n++) {
343 const double p =
S[n] -
mean;
353 double *prior,
double *prior_band_excit,
int track_noise)
356 const double *abs_var = dnch->
abs_var;
358 const double rratio = 1. - ratio;
359 const int *bin2band =
s->bin2band;
366 double *gain = dnch->
gain;
368 for (
int i = 0;
i <
s->bin_count;
i++) {
369 double sqr_new_gain, new_gain,
power, mag, mag_abs_var, new_mag_abs_var;
373 noisy_data[
i] = mag =
hypot(fft_data_flt[
i].re, fft_data_flt[
i].im);
376 noisy_data[
i] = mag =
hypot(fft_data_dbl[
i].re, fft_data_dbl[
i].im);
381 mag_abs_var =
power / abs_var[
i];
382 new_mag_abs_var = ratio * prior[
i] + rratio *
fmax(mag_abs_var - 1.0, 0.0);
383 new_gain = new_mag_abs_var / (1.0 + new_mag_abs_var);
384 sqr_new_gain = new_gain * new_gain;
385 prior[
i] = mag_abs_var * sqr_new_gain;
391 double flatness, num, den;
395 flatness = num / den;
396 if (flatness > 0.8) {
398 const double new_floor =
av_clipd(10.0 * log10(den) - 100.0 +
offset, -90., -20.);
405 for (
int i = 0;
i <
s->number_of_bands;
i++) {
410 for (
int i = 0;
i <
s->bin_count;
i++)
413 for (
int i = 0;
i <
s->number_of_bands;
i++) {
414 band_excit[
i] =
fmax(band_excit[
i],
415 s->band_alpha[
i] * band_excit[
i] +
416 s->band_beta[
i] * prior_band_excit[
i]);
417 prior_band_excit[
i] = band_excit[
i];
420 for (
int j = 0,
i = 0; j <
s->number_of_bands; j++) {
421 for (
int k = 0; k <
s->number_of_bands; k++) {
426 for (
int i = 0;
i <
s->bin_count;
i++)
427 dnch->
amt[
i] = band_amt[bin2band[
i]];
429 for (
int i = 0;
i <
s->bin_count;
i++) {
430 if (dnch->
amt[
i] > abs_var[
i]) {
433 const double limit = sqrt(abs_var[
i] / dnch->
amt[
i]);
441 memcpy(smoothed_gain, gain,
s->bin_count *
sizeof(*smoothed_gain));
442 if (
s->gain_smooth > 0) {
443 const int r =
s->gain_smooth;
445 for (
int i =
r;
i <
s->bin_count -
r;
i++) {
446 const double gc = gain[
i];
447 double num = 0., den = 0.;
449 for (
int j = -
r; j <=
r; j++) {
450 const double g = gain[
i + j];
451 const double d = 1. -
fabs(
g - gc);
457 smoothed_gain[
i] = num / den;
463 for (
int i = 0;
i <
s->bin_count;
i++) {
464 const float new_gain = smoothed_gain[
i];
466 fft_data_flt[
i].
re *= new_gain;
467 fft_data_flt[
i].
im *= new_gain;
471 for (
int i = 0;
i <
s->bin_count;
i++) {
472 const double new_gain = smoothed_gain[
i];
474 fft_data_dbl[
i].
re *= new_gain;
475 fft_data_dbl[
i].
im *= new_gain;
483 double d = x / 7500.0;
485 return 13.0 * atan(7.6
E-4 * x) + 3.5 * atan(
d *
d);
491 return lrint(
s->band_centre[0] / 1.5);
493 return s->band_centre[band];
503 i =
lrint(
s->band_centre[band] / 1.224745);
506 return FFMIN(
i,
s->sample_rate / 2);
512 double band_noise, d2, d3, d4, d5;
513 int i = 0, j = 0, k = 0;
517 for (
int m = j; m <
s->bin_count; m++) {
532 dnch->
rel_var[m] =
exp((d5 * d3 + band_noise * d4) *
C);
542 char *custom_noise_str, *p, *
arg, *saveptr =
NULL;
546 if (!
s->band_noise_str)
549 custom_noise_str = p =
av_strdup(
s->band_noise_str);
571 memcpy(dnch->
band_noise, band_noise,
sizeof(band_noise));
579 if (
s->track_residual)
583 if (update_auto_var) {
588 if (
s->track_residual) {
607 for (
int i = 0;
i <
s->bin_count;
i++) {
619 mean += band_noise[
i];
623 band_noise[
i] -=
mean;
630 double wscale, sar, sum, sdiv;
631 int i, j, k, m, n,
ret, tx_type;
640 s->sample_size =
sizeof(
float);
646 s->sample_size =
sizeof(
double);
657 s->channels =
inlink->ch_layout.nb_channels;
658 s->sample_rate =
inlink->sample_rate;
659 s->sample_advance =
s->sample_rate / 80;
660 s->window_length = 3 *
s->sample_advance;
661 s->fft_length2 = 1 << (32 -
ff_clz(
s->window_length));
662 s->fft_length =
s->fft_length2;
663 s->buffer_length =
s->fft_length * 2;
664 s->bin_count =
s->fft_length2 / 2 + 1;
666 s->band_centre[0] = 80;
668 s->band_centre[
i] =
lrint(1.5 *
s->band_centre[
i - 1] + 5.0);
669 if (
s->band_centre[
i] < 1000) {
670 s->band_centre[
i] = 10 * (
s->band_centre[
i] / 10);
671 }
else if (
s->band_centre[
i] < 5000) {
672 s->band_centre[
i] = 50 * ((
s->band_centre[
i] + 20) / 50);
673 }
else if (
s->band_centre[
i] < 15000) {
674 s->band_centre[
i] = 100 * ((
s->band_centre[
i] + 45) / 100);
676 s->band_centre[
i] = 1000 * ((
s->band_centre[
i] + 495) / 1000);
693 s->matrix_b[
i++] = pow(k, j);
698 s->matrix_c[
i++] = pow(j, k);
700 s->window =
av_calloc(
s->window_length,
sizeof(*
s->window));
701 s->bin2band =
av_calloc(
s->bin_count,
sizeof(*
s->bin2band));
702 if (!
s->window || !
s->bin2band)
705 sdiv =
s->band_multiplier;
706 for (
i = 0;
i <
s->bin_count;
i++)
709 s->number_of_bands =
s->bin2band[
s->bin_count - 1] + 1;
711 s->band_alpha =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_alpha));
712 s->band_beta =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_beta));
713 if (!
s->band_alpha || !
s->band_beta)
716 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
719 switch (
s->noise_type) {
786 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
792 p1 = pow(0.1, 2.5 / sdiv);
793 p2 = pow(0.1, 1.0 / sdiv);
795 for (m = 0; m <
s->number_of_bands; m++) {
796 for (n = 0; n <
s->number_of_bands; n++) {
807 for (m = 0; m <
s->number_of_bands; m++) {
809 prior_band_excit[m] = 0.0;
812 for (m = 0; m <
s->bin_count; m++)
816 for (m = 0; m <
s->number_of_bands; m++) {
817 for (n = 0; n <
s->number_of_bands; n++)
823 for (
int i = 0;
i <
s->number_of_bands;
i++) {
824 if (
i <
lrint(12.0 * sdiv)) {
827 dnch->
band_excit[
i] = pow(0.1, 2.5 - 0.2 * (
i / sdiv - 14.0));
832 for (
int i = 0;
i <
s->buffer_length;
i++)
836 for (
int i = 0;
i <
s->number_of_bands;
i++)
837 for (
int k = 0; k <
s->number_of_bands; k++)
842 sar =
s->sample_advance /
s->sample_rate;
843 for (
int i = 0;
i <
s->bin_count;
i++) {
844 if ((
i ==
s->fft_length2) || (
s->bin2band[
i] > j)) {
845 double d6 = (
i - 1) *
s->sample_rate /
s->fft_length;
846 double d7 =
fmin(0.008 + 2.2 / d6, 0.03);
847 s->band_alpha[j] =
exp(-sar / d7);
848 s->band_beta[j] = 1.0 -
s->band_alpha[j];
857 wscale = sqrt(8.0 / (9.0 *
s->fft_length));
859 for (
int i = 0;
i <
s->window_length;
i++) {
860 double d10 = sin(
i *
M_PI /
s->window_length);
866 s->window_weight = 0.5 * sum;
867 s->floor = (1LL << 48) *
exp(-23.025558369790467) *
s->window_weight;
868 s->sample_floor =
s->floor *
exp(4.144600506562284);
870 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
884 if (
s->noise_band_edge[j] >
lrint(1.1 *
s->noise_band_edge[j - 1]))
909 double mag2, var = 0.0, avr = 0.0, avi = 0.0;
912 double *fft_in_dbl = dnch->
fft_in;
913 float *fft_in_flt = dnch->
fft_in;
914 int edge, j, k, n, edgemax;
918 for (
int i = 0;
i <
s->window_length;
i++)
919 fft_in_flt[
i] =
s->window[
i] * src_flt[
i] * (1LL << 23);
921 for (
int i =
s->window_length; i < s->fft_length2;
i++)
925 for (
int i = 0;
i <
s->window_length;
i++)
926 fft_in_dbl[
i] =
s->window[
i] * src_dbl[
i] * (1LL << 23);
928 for (
int i =
s->window_length; i < s->fft_length2;
i++)
935 edge =
s->noise_band_edge[0];
940 for (
int i = j;
i <= edgemax;
i++) {
941 if ((
i == j) && (
i < edgemax)) {
950 j =
s->noise_band_edge[k];
961 avr += fft_out_flt[n].
re;
962 avi += fft_out_flt[n].
im;
963 mag2 = fft_out_flt[n].
re * fft_out_flt[n].
re +
964 fft_out_flt[n].
im * fft_out_flt[n].
im;
967 avr += fft_out_dbl[n].
re;
968 avi += fft_out_dbl[n].
im;
969 mag2 = fft_out_dbl[n].
re * fft_out_dbl[n].
re +
970 fft_out_dbl[n].
im * fft_out_dbl[n].
im;
974 mag2 =
fmax(mag2,
s->sample_floor);
988 double *sample_noise)
990 for (
int i = 0;
i <
s->noise_band_count;
i++) {
1001 sample_noise[
i] = sample_noise[
i - 1];
1007 double *sample_noise)
1014 temp[m] = sample_noise[m];
1019 sum +=
s->matrix_b[
i++] *
temp[n];
1020 s->vector_b[m] = sum;
1026 sum +=
s->matrix_c[
i++] *
s->vector_b[n];
1034 new_band_noise[m] =
temp[m];
1035 new_band_noise[m] =
av_clipd(new_band_noise[m], -24.0, 24.0);
1039 memcpy(dnch->
band_noise, new_band_noise,
sizeof(new_band_noise));
1048 const int window_length =
s->window_length;
1049 const double *
window =
s->window;
1051 for (
int ch = start; ch < end; ch++) {
1053 const double *src_dbl = (
const double *)in->
extended_data[ch];
1054 const float *src_flt = (
const float *)in->
extended_data[ch];
1056 double *fft_in_dbl = dnch->
fft_in;
1057 float *fft_in_flt = dnch->
fft_in;
1059 switch (
s->format) {
1061 for (
int m = 0; m < window_length; m++)
1062 fft_in_flt[m] =
window[m] * src_flt[m] * (1LL << 23);
1064 for (
int m = window_length; m <
s->fft_length2; m++)
1065 fft_in_flt[m] = 0.
f;
1068 for (
int m = 0; m < window_length; m++)
1069 fft_in_dbl[m] =
window[m] * src_dbl[m] * (1LL << 23);
1071 for (
int m = window_length; m <
s->fft_length2; m++)
1085 switch (
s->format) {
1087 for (
int m = 0; m < window_length; m++)
1088 dst[m] +=
s->window[m] * fft_in_flt[m] / (1LL << 23);
1091 for (
int m = 0; m < window_length; m++)
1092 dst[m] +=
s->window[m] * fft_in_dbl[m] / (1LL << 23);
1105 const int output_mode =
ctx->is_disabled ?
IN_MODE :
s->output_mode;
1106 const int offset =
s->window_length -
s->sample_advance;
1109 for (
int ch = 0; ch <
s->channels; ch++) {
1110 uint8_t *
src = (uint8_t *)
s->winframe->extended_data[ch];
1112 memmove(
src,
src +
s->sample_advance *
s->sample_size,
1117 (
s->sample_advance - in->
nb_samples) *
s->sample_size);
1120 if (
s->track_noise) {
1121 double average = 0.0,
min = DBL_MAX,
max = -DBL_MAX;
1123 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1131 average /=
inlink->ch_layout.nb_channels;
1133 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1136 switch (
s->noise_floor_link) {
1151 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1157 s->sample_noise = 1;
1158 s->sample_noise_blocks = 0;
1161 if (
s->sample_noise) {
1162 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1167 s->sample_noise_blocks++;
1171 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1175 if (
s->sample_noise_blocks <= 0)
1181 s->sample_noise = 0;
1182 s->sample_noise_blocks = 0;
1201 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1204 const double *orig_dbl = (
const double *)
s->winframe->extended_data[ch];
1205 const float *orig_flt = (
const float *)
s->winframe->extended_data[ch];
1206 double *dst_dbl = (
double *)
out->extended_data[ch];
1207 float *dst_flt = (
float *)
out->extended_data[ch];
1209 switch (output_mode) {
1211 switch (
s->format) {
1213 for (
int m = 0; m <
out->nb_samples; m++)
1214 dst_flt[m] = orig_flt[m];
1217 for (
int m = 0; m <
out->nb_samples; m++)
1218 dst_dbl[m] = orig_dbl[m];
1223 switch (
s->format) {
1225 for (
int m = 0; m <
out->nb_samples; m++)
1226 dst_flt[m] =
src[m];
1229 for (
int m = 0; m <
out->nb_samples; m++)
1230 dst_dbl[m] =
src[m];
1235 switch (
s->format) {
1237 for (
int m = 0; m <
out->nb_samples; m++)
1238 dst_flt[m] = orig_flt[m] -
src[m];
1241 for (
int m = 0; m <
out->nb_samples; m++)
1242 dst_dbl[m] = orig_dbl[m] -
src[m];
1253 memmove(
src,
src +
s->sample_advance, (
s->window_length -
s->sample_advance) *
sizeof(*
src));
1254 memset(
src + (
s->window_length -
s->sample_advance), 0,
s->sample_advance *
sizeof(*
src));
1300 for (
int ch = 0; ch <
s->channels; ch++) {
1326 char *res,
int res_len,
int flags)
1335 if (!strcmp(cmd,
"sample_noise") || !strcmp(cmd,
"sn"))
1338 for (
int ch = 0; ch <
s->channels; ch++) {
1363 .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)
const AVFilterPad ff_audio_default_filterpad[1]
An AVFilterPad array whose only entry has name "default" and is of type AVMEDIA_TYPE_AUDIO.
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 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.