Go to the documentation of this file.
32 #define C (M_LN10 * 0.1)
33 #define SOLVE_SIZE (5)
34 #define NB_PROFILE_BANDS (15)
158 #define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x)
159 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
160 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
225 d1 =
a /
s->band_centre[band];
226 d1 = 10.0 * log(1.0 + d1 * d1) /
M_LN10;
227 d2 =
b /
s->band_centre[band];
228 d2 = 10.0 * log(1.0 + d2 * d2) /
M_LN10;
229 d3 =
s->band_centre[band] /
c;
230 d3 = 10.0 * log(1.0 + d3 * d3) /
M_LN10;
232 return -d1 + d2 - d3;
237 for (
int i = 0;
i <
size - 1;
i++) {
238 for (
int j =
i + 1; j <
size; j++) {
242 for (
int k =
i + 1; k <
size; k++) {
251 for (
int i = 0;
i <
size - 1;
i++) {
252 for (
int j =
i + 1; j <
size; j++) {
254 vector[j] -=
d * vector[
i];
260 for (
int i =
size - 2;
i >= 0;
i--) {
261 double d = vector[
i];
262 for (
int j =
i + 1; j <
size; j++)
272 double product, sum,
f;
282 s->vector_b[j] = sum;
287 f = 15.0 + log(
f / 1.5) / log(1.5);
291 sum += product *
s->vector_b[j];
301 return (
b *
a - 1.0) / (
b +
a - 2.0);
303 return (
b *
a - 2.0 *
a + 1.0) / (
b -
a);
308 double floor,
int len,
double *rnum,
double *rden)
310 double num = 0., den = 0.;
313 for (
int n = 0; n <
len; n++) {
314 const double v = spectral[n];
339 for (
int n = 0; n <
size; n++) {
340 const double p =
S[n] -
mean;
351 double *prior,
double *prior_band_excit,
int track_noise)
354 const double *abs_var = dnch->
abs_var;
356 const double rratio = 1. - ratio;
357 const int *bin2band =
s->bin2band;
362 double *gain = dnch->
gain;
364 for (
int i = 0;
i <
s->bin_count;
i++) {
365 double sqr_new_gain, new_gain,
power, mag, mag_abs_var, new_mag_abs_var;
367 noisy_data[
i] = mag =
hypot(fft_data[
i].
re, fft_data[
i].
im);
369 mag_abs_var =
power / abs_var[
i];
370 new_mag_abs_var = ratio * prior[
i] + rratio *
fmax(mag_abs_var - 1.0, 0.0);
371 new_gain = new_mag_abs_var / (1.0 + new_mag_abs_var);
372 sqr_new_gain = new_gain * new_gain;
373 prior[
i] = mag_abs_var * sqr_new_gain;
379 double flatness, num, den;
383 flatness = num / den;
384 if (flatness > 0.8) {
386 const double new_floor =
av_clipd(10.0 * log10(den) - 100.0 +
offset, -90., -20.);
393 for (
int i = 0;
i <
s->number_of_bands;
i++) {
398 for (
int i = 0;
i <
s->bin_count;
i++)
401 for (
int i = 0;
i <
s->number_of_bands;
i++) {
402 band_excit[
i] =
fmax(band_excit[
i],
403 s->band_alpha[
i] * band_excit[
i] +
404 s->band_beta[
i] * prior_band_excit[
i]);
405 prior_band_excit[
i] = band_excit[
i];
408 for (
int j = 0,
i = 0; j <
s->number_of_bands; j++) {
409 for (
int k = 0; k <
s->number_of_bands; k++) {
414 for (
int i = 0;
i <
s->bin_count;
i++)
415 dnch->
amt[
i] = band_amt[bin2band[
i]];
417 for (
int i = 0;
i <
s->bin_count;
i++) {
418 if (dnch->
amt[
i] > abs_var[
i]) {
421 const double limit = sqrt(abs_var[
i] / dnch->
amt[
i]);
429 memcpy(smoothed_gain, gain,
s->bin_count *
sizeof(*smoothed_gain));
430 if (
s->gain_smooth > 0) {
431 const int r =
s->gain_smooth;
433 for (
int i =
r;
i <
s->bin_count -
r;
i++) {
434 const double gc = gain[
i];
435 double num = 0., den = 0.;
437 for (
int j = -
r; j <=
r; j++) {
438 const double g = gain[
i + j];
439 const double d = 1. -
fabs(
g - gc);
445 smoothed_gain[
i] = num / den;
449 for (
int i = 0;
i <
s->bin_count;
i++) {
450 const double new_gain = smoothed_gain[
i];
452 fft_data[
i].
re *= new_gain;
453 fft_data[
i].
im *= new_gain;
459 double d = x / 7500.0;
461 return 13.0 * atan(7.6
E-4 * x) + 3.5 * atan(
d *
d);
467 return lrint(
s->band_centre[0] / 1.5);
469 return s->band_centre[band];
479 i =
lrint(
s->band_centre[band] / 1.224745);
482 return FFMIN(
i,
s->sample_rate / 2);
488 double band_noise, d2, d3, d4, d5;
489 int i = 0, j = 0, k = 0;
493 for (
int m = j; m <
s->bin_count; m++) {
508 dnch->
rel_var[m] =
exp((d5 * d3 + band_noise * d4) *
C);
518 char *custom_noise_str, *p, *
arg, *saveptr =
NULL;
522 if (!
s->band_noise_str)
525 custom_noise_str = p =
av_strdup(
s->band_noise_str);
547 memcpy(dnch->
band_noise, band_noise,
sizeof(band_noise));
555 if (
s->track_residual)
559 if (update_auto_var) {
564 if (
s->track_residual) {
583 for (
int i = 0;
i <
s->bin_count;
i++) {
595 mean += band_noise[
i];
599 band_noise[
i] -=
mean;
606 double wscale, sar, sum, sdiv;
607 int i, j, k, m, n,
ret;
613 s->channels =
inlink->ch_layout.nb_channels;
614 s->sample_rate =
inlink->sample_rate;
615 s->sample_advance =
s->sample_rate / 80;
616 s->window_length = 3 *
s->sample_advance;
617 s->fft_length2 = 1 << (32 -
ff_clz(
s->window_length));
618 s->fft_length =
s->fft_length2;
619 s->buffer_length =
s->fft_length * 2;
620 s->bin_count =
s->fft_length2 / 2 + 1;
622 s->band_centre[0] = 80;
624 s->band_centre[
i] =
lrint(1.5 *
s->band_centre[
i - 1] + 5.0);
625 if (
s->band_centre[
i] < 1000) {
626 s->band_centre[
i] = 10 * (
s->band_centre[
i] / 10);
627 }
else if (
s->band_centre[
i] < 5000) {
628 s->band_centre[
i] = 50 * ((
s->band_centre[
i] + 20) / 50);
629 }
else if (
s->band_centre[
i] < 15000) {
630 s->band_centre[
i] = 100 * ((
s->band_centre[
i] + 45) / 100);
632 s->band_centre[
i] = 1000 * ((
s->band_centre[
i] + 495) / 1000);
649 s->matrix_b[
i++] = pow(k, j);
654 s->matrix_c[
i++] = pow(j, k);
656 s->window =
av_calloc(
s->window_length,
sizeof(*
s->window));
657 s->bin2band =
av_calloc(
s->bin_count,
sizeof(*
s->bin2band));
658 if (!
s->window || !
s->bin2band)
661 sdiv =
s->band_multiplier;
662 for (
i = 0;
i <
s->bin_count;
i++)
665 s->number_of_bands =
s->bin2band[
s->bin_count - 1] + 1;
667 s->band_alpha =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_alpha));
668 s->band_beta =
av_calloc(
s->number_of_bands,
sizeof(*
s->band_beta));
669 if (!
s->band_alpha || !
s->band_beta)
672 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
676 switch (
s->noise_type) {
743 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
749 p1 = pow(0.1, 2.5 / sdiv);
750 p2 = pow(0.1, 1.0 / sdiv);
752 for (m = 0; m <
s->number_of_bands; m++) {
753 for (n = 0; n <
s->number_of_bands; n++) {
764 for (m = 0; m <
s->number_of_bands; m++) {
766 prior_band_excit[m] = 0.0;
769 for (m = 0; m <
s->bin_count; m++)
773 for (m = 0; m <
s->number_of_bands; m++) {
774 for (n = 0; n <
s->number_of_bands; n++)
780 for (
int i = 0;
i <
s->number_of_bands;
i++) {
781 if (
i <
lrint(12.0 * sdiv)) {
784 dnch->
band_excit[
i] = pow(0.1, 2.5 - 0.2 * (
i / sdiv - 14.0));
789 for (
int i = 0;
i <
s->buffer_length;
i++)
793 for (
int i = 0;
i <
s->number_of_bands;
i++)
794 for (
int k = 0; k <
s->number_of_bands; k++)
799 sar =
s->sample_advance /
s->sample_rate;
800 for (
int i = 0;
i <
s->bin_count;
i++) {
801 if ((
i ==
s->fft_length2) || (
s->bin2band[
i] > j)) {
802 double d6 = (
i - 1) *
s->sample_rate /
s->fft_length;
803 double d7 =
fmin(0.008 + 2.2 / d6, 0.03);
804 s->band_alpha[j] =
exp(-sar / d7);
805 s->band_beta[j] = 1.0 -
s->band_alpha[j];
814 wscale = sqrt(8.0 / (9.0 *
s->fft_length));
816 for (
int i = 0;
i <
s->window_length;
i++) {
817 double d10 = sin(
i *
M_PI /
s->window_length);
823 s->window_weight = 0.5 * sum;
824 s->floor = (1LL << 48) *
exp(-23.025558369790467) *
s->window_weight;
825 s->sample_floor =
s->floor *
exp(4.144600506562284);
827 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
841 if (
s->noise_band_edge[j] >
lrint(1.1 *
s->noise_band_edge[j - 1]))
865 double mag2, var = 0.0, avr = 0.0, avi = 0.0;
866 int edge, j, k, n, edgemax;
868 for (
int i = 0;
i <
s->window_length;
i++)
871 for (
int i =
s->window_length; i < s->fft_length2;
i++)
876 edge =
s->noise_band_edge[0];
881 for (
int i = j;
i <= edgemax;
i++) {
882 if ((
i == j) && (
i < edgemax)) {
891 j =
s->noise_band_edge[k];
904 mag2 =
fmax(mag2,
s->sample_floor);
918 double *sample_noise)
920 for (
int i = 0;
i <
s->noise_band_count;
i++) {
931 sample_noise[
i] = sample_noise[
i - 1];
937 double *sample_noise)
944 temp[m] = sample_noise[m];
949 sum +=
s->matrix_b[
i++] *
temp[n];
950 s->vector_b[m] = sum;
956 sum +=
s->matrix_c[
i++] *
s->vector_b[n];
964 new_band_noise[m] =
temp[m];
965 new_band_noise[m] =
av_clipd(new_band_noise[m], -24.0, 24.0);
969 memcpy(dnch->
band_noise, new_band_noise,
sizeof(new_band_noise));
978 const int window_length =
s->window_length;
979 const double *
window =
s->window;
981 for (
int ch = start; ch < end; ch++) {
985 float *fft_in = dnch->
fft_in;
987 for (
int m = 0; m < window_length; m++)
988 fft_in[m] =
window[m] *
src[m] * (1LL << 23);
990 for (
int m = window_length; m <
s->fft_length2; m++)
1002 for (
int m = 0; m < window_length; m++)
1003 dst[m] +=
s->window[m] * fft_in[m] / (1LL << 23);
1014 const int output_mode =
ctx->is_disabled ?
IN_MODE :
s->output_mode;
1015 const int offset =
s->window_length -
s->sample_advance;
1018 for (
int ch = 0; ch <
s->channels; ch++) {
1019 float *
src = (
float *)
s->winframe->extended_data[ch];
1021 memmove(
src, &
src[
s->sample_advance],
offset *
sizeof(
float));
1026 if (
s->track_noise) {
1027 double average = 0.0,
min = DBL_MAX,
max = -DBL_MAX;
1029 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1037 average /=
inlink->ch_layout.nb_channels;
1039 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1042 switch (
s->noise_floor_link) {
1057 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1063 s->sample_noise = 1;
1064 s->sample_noise_blocks = 0;
1067 if (
s->sample_noise) {
1068 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1073 s->sample_noise_blocks++;
1077 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1081 if (
s->sample_noise_blocks <= 0)
1087 s->sample_noise = 0;
1088 s->sample_noise_blocks = 0;
1107 for (
int ch = 0; ch <
inlink->ch_layout.nb_channels; ch++) {
1110 const float *orig = (
const float *)
s->winframe->extended_data[ch];
1111 float *dst = (
float *)
out->extended_data[ch];
1113 switch (output_mode) {
1115 for (
int m = 0; m <
out->nb_samples; m++)
1119 for (
int m = 0; m <
out->nb_samples; m++)
1123 for (
int m = 0; m <
out->nb_samples; m++)
1124 dst[m] = orig[m] -
src[m];
1132 memmove(
src,
src +
s->sample_advance, (
s->window_length -
s->sample_advance) *
sizeof(*
src));
1133 memset(
src + (
s->window_length -
s->sample_advance), 0,
s->sample_advance *
sizeof(*
src));
1179 for (
int ch = 0; ch <
s->channels; ch++) {
1205 char *res,
int res_len,
int flags)
1214 if (!strcmp(cmd,
"sample_noise") || !strcmp(cmd,
"sn"))
1217 for (
int ch = 0; ch <
s->channels; ch++) {
1249 .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
static void process_frame(AVFilterContext *ctx, AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, AVComplexFloat *fft_data, double *prior, double *prior_band_excit, int track_noise)
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[]
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
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.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
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
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.
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)
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
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.