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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];
231 d2 =
b /
s->band_centre[band];
233 d3 =
s->band_centre[band] /
c;
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;
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);
384 mag_abs_var =
power / abs_var[
i];
385 new_mag_abs_var = ratio * prior[
i] + rratio *
fmax(mag_abs_var - 1.0, 0.0);
386 new_gain = new_mag_abs_var / (1.0 + new_mag_abs_var);
387 sqr_new_gain = new_gain * new_gain;
388 prior[
i] = mag_abs_var * sqr_new_gain;
394 double flatness, num, den;
398 flatness = num / den;
399 if (flatness > 0.8) {
401 const double new_floor =
av_clipd(10.0 * log10(den) - 100.0 +
offset, -90., -20.);
408 for (
int i = 0;
i <
s->number_of_bands;
i++) {
413 for (
int i = 0;
i <
s->bin_count;
i++)
416 for (
int i = 0;
i <
s->number_of_bands;
i++) {
417 band_excit[
i] =
fmax(band_excit[
i],
418 s->band_alpha[
i] * band_excit[
i] +
419 s->band_beta[
i] * prior_band_excit[
i]);
420 prior_band_excit[
i] = band_excit[
i];
423 for (
int j = 0,
i = 0; j <
s->number_of_bands; j++) {
424 for (
int k = 0; k <
s->number_of_bands; k++) {
429 for (
int i = 0;
i <
s->bin_count;
i++)
430 dnch->
amt[
i] = band_amt[bin2band[
i]];
432 for (
int i = 0;
i <
s->bin_count;
i++) {
433 if (dnch->
amt[
i] > abs_var[
i]) {
436 const double limit = sqrt(abs_var[
i] / dnch->
amt[
i]);
444 memcpy(smoothed_gain, gain,
s->bin_count *
sizeof(*smoothed_gain));
445 if (
s->gain_smooth > 0) {
446 const int r =
s->gain_smooth;
448 for (
int i =
r;
i <
s->bin_count -
r;
i++) {
449 const double gc = gain[
i];
450 double num = 0., den = 0.;
452 for (
int j = -
r; j <=
r; j++) {
453 const double g = gain[
i + j];
454 const double d = 1. -
fabs(
g - gc);
460 smoothed_gain[
i] = num / den;
466 for (
int i = 0;
i <
s->bin_count;
i++) {
467 const float new_gain = smoothed_gain[
i];
469 fft_data_flt[
i].
re *= new_gain;
470 fft_data_flt[
i].
im *= new_gain;
474 for (
int i = 0;
i <
s->bin_count;
i++) {
475 const double new_gain = smoothed_gain[
i];
477 fft_data_dbl[
i].
re *= new_gain;
478 fft_data_dbl[
i].
im *= new_gain;
486 double d = x / 7500.0;
488 return 13.0 * atan(7.6
E-4 * x) + 3.5 * atan(
d *
d);
494 return lrint(
s->band_centre[0] / 1.5);
496 return s->band_centre[band];
506 i =
lrint(
s->band_centre[band] / 1.224745);
509 return FFMIN(
i,
s->sample_rate / 2);
515 double band_noise, d2, d3, d4, d5;
516 int i = 0, j = 0, k = 0;
520 for (
int m = j; m <
s->bin_count; m++) {
535 dnch->
rel_var[m] =
exp((d5 * d3 + band_noise * d4) *
C);
545 char *custom_noise_str, *p, *
arg, *saveptr =
NULL;
549 if (!
s->band_noise_str)
552 custom_noise_str = p =
av_strdup(
s->band_noise_str);
574 memcpy(dnch->
band_noise, band_noise,
sizeof(band_noise));
582 if (
s->track_residual)
586 if (update_auto_var) {
591 if (
s->track_residual) {
610 for (
int i = 0;
i <
s->bin_count;
i++) {
622 mean += band_noise[
i];
626 band_noise[
i] -=
mean;
633 double wscale, sar, sum, sdiv;
634 int i, j, k, m, n,
ret, tx_type;
643 s->sample_size =
sizeof(
float);
649 s->sample_size =
sizeof(
double);
660 s->channels =
inlink->ch_layout.nb_channels;
661 s->sample_rate =
inlink->sample_rate;
662 s->sample_advance =
s->sample_rate / 80;
663 s->window_length = 3 *
s->sample_advance;
664 s->fft_length2 = 1 << (32 -
ff_clz(
s->window_length));
665 s->fft_length =
s->fft_length2;
666 s->buffer_length =
s->fft_length * 2;
667 s->bin_count =
s->fft_length2 / 2 + 1;
669 s->band_centre[0] = 80;
671 s->band_centre[
i] =
lrint(1.5 *
s->band_centre[
i - 1] + 5.0);
672 if (
s->band_centre[
i] < 1000) {
673 s->band_centre[
i] = 10 * (
s->band_centre[
i] / 10);
674 }
else if (
s->band_centre[
i] < 5000) {
675 s->band_centre[
i] = 50 * ((
s->band_centre[
i] + 20) / 50);
676 }
else if (
s->band_centre[
i] < 15000) {
677 s->band_centre[
i] = 100 * ((
s->band_centre[
i] + 45) / 100);
679 s->band_centre[
i] = 1000 * ((
s->band_centre[
i] + 495) / 1000);
696 s->matrix_b[
i++] = pow(k, j);
701 s->matrix_c[
i++] = pow(j, k);
703 s->window =
av_calloc(
s->window_length,
sizeof(*
s->window));
704 s->bin2band =
av_calloc(
s->bin_count,
sizeof(*
s->bin2band));
705 if (!
s->window || !
s->bin2band)
708 sdiv =
s->band_multiplier;
709 for (
i = 0;
i <
s->bin_count;
i++)
712 s->number_of_bands =
s->bin2band[
s->bin_count - 1] + 1;
714 s->band_alpha =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_alpha));
715 s->band_beta =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_beta));
716 if (!
s->band_alpha || !
s->band_beta)
719 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
722 switch (
s->noise_type) {
789 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
795 p1 = pow(0.1, 2.5 / sdiv);
796 p2 = pow(0.1, 1.0 / sdiv);
798 for (m = 0; m <
s->number_of_bands; m++) {
799 for (n = 0; n <
s->number_of_bands; n++) {
810 for (m = 0; m <
s->number_of_bands; m++) {
812 prior_band_excit[m] = 0.0;
815 for (m = 0; m <
s->bin_count; m++)
819 for (m = 0; m <
s->number_of_bands; m++) {
820 for (n = 0; n <
s->number_of_bands; n++)
826 for (
int i = 0;
i <
s->number_of_bands;
i++) {
827 if (
i <
lrint(12.0 * sdiv)) {
830 dnch->
band_excit[
i] = pow(0.1, 2.5 - 0.2 * (
i / sdiv - 14.0));
835 for (
int i = 0;
i <
s->buffer_length;
i++)
839 for (
int i = 0;
i <
s->number_of_bands;
i++)
840 for (
int k = 0; k <
s->number_of_bands; k++)
845 sar =
s->sample_advance /
s->sample_rate;
846 for (
int i = 0;
i <
s->bin_count;
i++) {
847 if ((
i ==
s->fft_length2) || (
s->bin2band[
i] > j)) {
848 double d6 = (
i - 1) *
s->sample_rate /
s->fft_length;
849 double d7 =
fmin(0.008 + 2.2 / d6, 0.03);
850 s->band_alpha[j] =
exp(-sar / d7);
851 s->band_beta[j] = 1.0 -
s->band_alpha[j];
860 wscale = sqrt(8.0 / (9.0 *
s->fft_length));
862 for (
int i = 0;
i <
s->window_length;
i++) {
863 double d10 = sin(
i *
M_PI /
s->window_length);
869 s->window_weight = 0.5 * sum;
870 s->floor = (1LL << 48) *
exp(-23.025558369790467) *
s->window_weight;
871 s->sample_floor =
s->floor *
exp(4.144600506562284);
873 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
887 if (
s->noise_band_edge[j] >
lrint(1.1 *
s->noise_band_edge[j - 1]))
912 double mag2, var = 0.0, avr = 0.0, avi = 0.0;
915 double *fft_in_dbl = dnch->
fft_in;
916 float *fft_in_flt = dnch->
fft_in;
917 int edge, j, k, n, edgemax;
921 for (
int i = 0;
i <
s->window_length;
i++)
922 fft_in_flt[
i] =
s->window[
i] * src_flt[
i] * (1LL << 23);
924 for (
int i =
s->window_length; i < s->fft_length2;
i++)
928 for (
int i = 0;
i <
s->window_length;
i++)
929 fft_in_dbl[
i] =
s->window[
i] * src_dbl[
i] * (1LL << 23);
931 for (
int i =
s->window_length; i < s->fft_length2;
i++)
938 edge =
s->noise_band_edge[0];
943 for (
int i = j;
i <= edgemax;
i++) {
944 if ((
i == j) && (
i < edgemax)) {
953 j =
s->noise_band_edge[k];
964 avr += fft_out_flt[n].
re;
965 avi += fft_out_flt[n].
im;
966 mag2 = fft_out_flt[n].
re * fft_out_flt[n].
re +
967 fft_out_flt[n].
im * fft_out_flt[n].
im;
970 avr += fft_out_dbl[n].
re;
971 avi += fft_out_dbl[n].
im;
972 mag2 = fft_out_dbl[n].
re * fft_out_dbl[n].
re +
973 fft_out_dbl[n].
im * fft_out_dbl[n].
im;
979 mag2 =
fmax(mag2,
s->sample_floor);
993 double *sample_noise)
995 for (
int i = 0;
i <
s->noise_band_count;
i++) {
1006 sample_noise[
i] = sample_noise[
i - 1];
1012 double *sample_noise)
1019 temp[m] = sample_noise[m];
1024 sum +=
s->matrix_b[
i++] *
temp[n];
1025 s->vector_b[m] = sum;
1031 sum +=
s->matrix_c[
i++] *
s->vector_b[n];
1039 new_band_noise[m] =
temp[m];
1040 new_band_noise[m] =
av_clipd(new_band_noise[m], -24.0, 24.0);
1044 memcpy(dnch->
band_noise, new_band_noise,
sizeof(new_band_noise));
1053 const int window_length =
s->window_length;
1054 const double *
window =
s->window;
1056 for (
int ch = start; ch < end; ch++) {
1058 const double *src_dbl = (
const double *)in->
extended_data[ch];
1059 const float *src_flt = (
const float *)in->
extended_data[ch];
1061 double *fft_in_dbl = dnch->
fft_in;
1062 float *fft_in_flt = dnch->
fft_in;
1064 switch (
s->format) {
1066 for (
int m = 0; m < window_length; m++)
1067 fft_in_flt[m] =
window[m] * src_flt[m] * (1LL << 23);
1069 for (
int m = window_length; m <
s->fft_length2; m++)
1070 fft_in_flt[m] = 0.
f;
1073 for (
int m = 0; m < window_length; m++)
1074 fft_in_dbl[m] =
window[m] * src_dbl[m] * (1LL << 23);
1076 for (
int m = window_length; m <
s->fft_length2; m++)
1090 switch (
s->format) {
1092 for (
int m = 0; m < window_length; m++)
1093 dst[m] +=
s->window[m] * fft_in_flt[m] / (1LL << 23);
1096 for (
int m = 0; m < window_length; m++)
1097 dst[m] +=
s->window[m] * fft_in_dbl[m] / (1LL << 23);
1110 const int output_mode =
ctx->is_disabled ?
IN_MODE :
s->output_mode;
1111 const int offset =
s->window_length -
s->sample_advance;
1114 for (
int ch = 0; ch <
s->channels; ch++) {
1115 uint8_t *
src = (uint8_t *)
s->winframe->extended_data[ch];
1117 memmove(
src,
src +
s->sample_advance *
s->sample_size,
1122 (
s->sample_advance - in->
nb_samples) *
s->sample_size);
1125 if (
s->track_noise) {
1126 double average = 0.0,
min = DBL_MAX,
max = -DBL_MAX;
1128 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1136 average /=
inlink->ch_layout.nb_channels;
1138 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1141 switch (
s->noise_floor_link) {
1156 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1162 s->sample_noise = 1;
1163 s->sample_noise_blocks = 0;
1166 if (
s->sample_noise) {
1167 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1172 s->sample_noise_blocks++;
1176 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1180 if (
s->sample_noise_blocks <= 0)
1186 s->sample_noise = 0;
1187 s->sample_noise_blocks = 0;
1206 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1209 const double *orig_dbl = (
const double *)
s->winframe->extended_data[ch];
1210 const float *orig_flt = (
const float *)
s->winframe->extended_data[ch];
1211 double *dst_dbl = (
double *)
out->extended_data[ch];
1212 float *dst_flt = (
float *)
out->extended_data[ch];
1214 switch (output_mode) {
1216 switch (
s->format) {
1218 for (
int m = 0; m <
out->nb_samples; m++)
1219 dst_flt[m] = orig_flt[m];
1222 for (
int m = 0; m <
out->nb_samples; m++)
1223 dst_dbl[m] = orig_dbl[m];
1228 switch (
s->format) {
1230 for (
int m = 0; m <
out->nb_samples; m++)
1231 dst_flt[m] =
src[m];
1234 for (
int m = 0; m <
out->nb_samples; m++)
1235 dst_dbl[m] =
src[m];
1240 switch (
s->format) {
1242 for (
int m = 0; m <
out->nb_samples; m++)
1243 dst_flt[m] = orig_flt[m] -
src[m];
1246 for (
int m = 0; m <
out->nb_samples; m++)
1247 dst_dbl[m] = orig_dbl[m] -
src[m];
1258 memmove(
src,
src +
s->sample_advance, (
s->window_length -
s->sample_advance) *
sizeof(*
src));
1259 memset(
src + (
s->window_length -
s->sample_advance), 0,
s->sample_advance *
sizeof(*
src));
1305 for (
int ch = 0; ch <
s->channels; ch++) {
1331 char *res,
int res_len,
int flags)
1340 if (!strcmp(cmd,
"sample_noise") || !strcmp(cmd,
"sn"))
1343 for (
int ch = 0; ch <
s->channels; ch++) {
1368 .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)
static int noise(AVBSFContext *ctx, AVPacket *pkt)
double band_noise[NB_PROFILE_BANDS]
AVChannelLayout ch_layout
Channel layout of the audio data.
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
static void scale(int *out, const int *in, const int w, const int h, const int shift)
#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)
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
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 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.