33 #define C (M_LN10 * 0.1) 35 #define RRATIO (1.0 - RATIO) 133 int noise_band_edge[17];
143 #define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x) 144 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 145 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM 175 d1 = 10.0 * log(1.0 + d1 * d1) /
M_LN10;
177 d2 = 10.0 * log(1.0 + d2 * d2) /
M_LN10;
179 d3 = 10.0 * log(1.0 + d3 * d3) /
M_LN10;
181 return lrint(-d1 + d2 - d3);
186 for (
int i = 0;
i < size - 1;
i++) {
187 for (
int j =
i + 1; j <
size; j++) {
191 for (
int k =
i + 1; k <
size; k++) {
192 array[j + k *
size] -= d * array[
i + k *
size];
198 static void solve(
double *matrix,
double *vector,
int size)
200 for (
int i = 0;
i < size - 1;
i++) {
201 for (
int j =
i + 1; j <
size; j++) {
202 double d = matrix[j +
i *
size];
203 vector[j] -= d * vector[
i];
207 vector[size - 1] /= matrix[size * size - 1];
209 for (
int i = size - 2;
i >= 0;
i--) {
210 double d = vector[
i];
211 for (
int j =
i + 1; j <
size; j++)
212 d -= matrix[
i + j * size] * vector[j];
213 vector[
i] = d / matrix[
i +
i *
size];
221 double product, sum,
f;
227 for (
int j = 0; j < 5; j++) {
229 for (
int k = 0; k < 15; k++)
236 f = 15.0 + log(f / 1.5) / log(1.5);
239 for (
int j = 0; j < 5; j++) {
251 double d1 = 0.0, d2 = 1.0;
254 for (
int k = start; k < end; k++) {
262 }
else if (d2 < 1.0
E-100) {
271 d2 = log(d2) + 230.2585 *
i;
286 return (b * a - 1.0) / (b + a - 2.0);
288 return (b * a - 2.0 * a + 1.0) / (b -
a);
296 double d1, d2, d3,
gain;
299 d1 = fft_data[0].
re * fft_data[0].
re;
303 gain = d3 / (1.0 + d3);
304 gain *= (gain + M_PI_4 /
fmax(d2, 1.0
E-6));
305 prior[0] = (d2 *
gain);
311 d1 = fft_data[
i].
re * fft_data[
i].
re + fft_data[
i].
im * fft_data[
i].
im;
318 gain = d3 / (1.0 + d3);
319 gain *= (gain + M_PI_4 /
fmax(d2, 1.0
E-6));
320 prior[
i] = d2 *
gain;
325 d1 = fft_data[0].
im * fft_data[0].
im;
332 gain = d3 / (1.0 + d3);
333 gain *= gain + M_PI_4 /
fmax(d2, 1.0
E-6);
354 for (k = i1 - 1; k >= 0; k--) {
366 double sum = 0.0,
min,
max;
369 for (i = i1 - 1; i > k; i--) {
376 min = 3.0E-4 * i *
i;
378 min = 3.0E-4 * (8 * i - 16);
381 max = 2.0E-4 * i *
i;
383 max = 2.0E-4 * (4 * i - 4);
410 for (
int i = 0; i < 15; i++)
446 double limit = sqrt(dnch->
abs_var[0] / dnch->
amt[0]);
471 gain = dnch->
gain[0];
478 gain = dnch->
gain[
i];
487 double d = x / 7500.0;
489 return 13.0 * atan(7.6
E-4 * x) + 3.5 * atan(d * d);
517 int i = 0, j = 0, k = 0;
536 dnch->
rel_var[m] =
exp((d5 * d3 + band_noise * d4) *
C);
540 for (i = 0; i < 15; i++)
552 char *p, *
arg, *saveptr =
NULL;
562 for (i = 0; i < 15; i++) {
563 if (!(arg =
av_strtok(p,
"| ", &saveptr)))
568 ret =
av_sscanf(arg,
"%d", &band_noise[i]);
574 band_noise[
i] =
av_clip(band_noise[i], -24, 24);
578 memcpy(dnch->
band_noise, band_noise,
sizeof(band_noise));
603 for (
int ch = 0; ch < s->
channels; ch++) {
614 double wscale, sar, sum, sdiv;
632 for (i = 1; i < 15; i++) {
645 for (j = 0; j < 5; j++) {
646 for (k = 0; k < 5; k++) {
648 for (m = 0; m < 15; m++)
649 s->
matrix_a[j + k * 5] += pow(m, j + k);
656 for (j = 0; j < 5; j++)
657 for (k = 0; k < 15; k++)
661 for (j = 0; j < 15; j++)
662 for (k = 0; k < 5; k++)
681 for (
int ch = 0; ch < inlink->
channels; ch++) {
686 for (i = 0; i < 15; i++)
690 for (i = 0; i < 15; i++)
694 for (i = 0; i < 15; i++)
707 for (i = 0; i < 512; i++)
713 for (i = 0; i < 512; i += j) {
755 for (
int ch = 0; ch < inlink->
channels; ch++) {
762 p1 = pow(0.1, 2.5 / sdiv);
763 p2 = pow(0.1, 1.0 / sdiv);
779 prior_band_excit[m] = 0.0;
794 if (i <
lrint(12.0 * sdiv)) {
795 dnch->
band_excit[
i] = pow(0.1, 1.45 + 0.1 * i / sdiv);
797 dnch->
band_excit[
i] = pow(0.1, 2.5 - 0.2 * (i / sdiv - 14.0));
818 double d7 =
fmin(0.008 + 2.2 / d6, 0.03);
843 for (
int j = 1; j < 16; j++) {
860 double d1, d2, d3, d4, d5, d6, d7, d8, d9, d10;
871 for (i = 1; i < len / 4; i++) {
873 d2 = 0.5 * (in[
i].
re + in[k].
re);
874 d1 = 0.5 * (in[
i].
im - in[k].
im);
875 d4 = 0.5 * (in[
i].
im + in[k].
im);
876 d3 = 0.5 * (in[k].
re - in[
i].
re);
877 in[
i].
re = d2 + d9 * d4 + d6 * d3;
878 in[
i].
im = d1 + d9 * d3 - d6 * d4;
879 in[k].
re = d2 - d9 * d4 - d6 * d3;
880 in[k].
im = -d1 + d9 * d3 - d6 * d4;
882 d9 += d9 * d8 - d6 * d7;
883 d6 += d6 * d8 + d10 * d7;
887 in[0].
re = d2 + in[0].
im;
888 in[0].
im = d2 - in[0].
im;
893 double d1, d2, d3, d4, d5, d6, d7, d8, d9, d10;
903 for (i = 1; i < len / 4; i++) {
905 d2 = 0.5 * (in[
i].
re + in[k].
re);
906 d1 = 0.5 * (in[
i].
im - in[k].
im);
907 d4 = 0.5 * (in[
i].
re - in[k].
re);
908 d3 = 0.5 * (in[
i].
im + in[k].
im);
909 in[
i].
re = d2 - d9 * d3 - d6 * d4;
910 in[
i].
im = d1 + d9 * d4 - d6 * d3;
911 in[k].
re = d2 + d9 * d3 + d6 * d4;
912 in[k].
im = -d1 + d9 * d4 - d6 * d3;
914 d9 += d9 * d8 - d6 * d7;
915 d6 += d6 * d8 + d10 * d7;
918 in[0].
re = 0.5 * (d2 + in[0].
im);
919 in[0].
im = 0.5 * (d2 - in[0].
im);
924 for (
int i = 0;
i < 15;
i++) {
937 double mag2, var = 0.0, avr = 0.0, avi = 0.0;
938 int edge, j, k, n, edgemax;
964 for (
int i = j;
i <= edgemax;
i++) {
965 if ((
i == j) && (
i < edgemax)) {
1002 double *sample_noise)
1015 sample_noise[
i] = sample_noise[
i - 1];
1021 double *sample_noise,
1024 int new_band_noise[15];
1026 double sum = 0.0, d1;
1027 float new_noise_floor;
1030 for (
int m = 0; m < 15; m++)
1031 temp[m] = sample_noise[m];
1035 for (
int m = 0; m < 5; m++) {
1037 for (n = 0; n < 15; n++)
1043 for (
int m = 0; m < 15; m++) {
1045 for (n = 0; n < 5; n++)
1052 for (
int m = 0; m < 15; m++)
1055 d1 = (
int)(sum / 15.0 - 0.5);
1057 i =
lrint(temp[7] - d1);
1059 for (d1 -= dnch->
band_noise[7] - i; d1 > -20.0; d1 -= 1.0)
1062 for (
int m = 0; m < 15; m++)
1065 new_noise_floor = d1 + 2.5;
1069 for (
int m = 0; m < 15; m++) {
1070 new_band_noise[m] =
lrint(temp[m]);
1071 new_band_noise[m] =
av_clip(new_band_noise[m], -24, 24);
1075 memcpy(dnch->
band_noise, new_band_noise,
sizeof(new_band_noise));
1091 const int start = (in->
channels * jobnr) / nb_jobs;
1092 const int end = (in->
channels * (jobnr+1)) / nb_jobs;
1094 for (
int ch = start; ch < end; ch++) {
1149 levels[
i] = levels[
i - 1];
1152 for (
int i = 0;
i < 15;
i++) {
1176 for (
int ch = 0; ch < inlink->
channels; ch++) {
1189 for (
int ch = 0; ch < inlink->
channels; ch++) {
1199 for (
int ch = 0; ch < inlink->
channels; ch++) {
1207 for (
int ch = 0; ch < inlink->
channels; ch++) {
1209 double sample_noise[15];
1230 for (
int ch = 0; ch < inlink->
channels; ch++) {
1233 float *orig = (
float *)in->extended_data[ch];
1247 dst[m] = orig[m] - src[m];
1318 for (
int ch = 0; ch < s->
channels; ch++) {
1375 char *res,
int res_len,
int flags)
1381 if (!strcmp(cmd,
"sample_noise") ||
1382 !strcmp(cmd,
"sn")) {
1383 if (!strcmp(args,
"start")) {
1386 }
else if (!strcmp(args,
"end") ||
1387 !strcmp(args,
"stop")) {
1426 .priv_class = &afftdn_class,
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
double noise_band_var[15]
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
double noise_band_norm[15]
This structure describes decoded (raw) audio or video data.
av_cold void av_fft_end(FFTContext *s)
Main libavfilter public API header.
static void set_parameters(AudioFFTDeNoiseContext *s)
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
FF_FILTER_FORWARD_STATUS(inlink, outlink)
static const AVOption afftdn_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 set_noise_profile(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *sample_noise, int new_profile)
void av_fft_permute(FFTContext *s, FFTComplex *z)
Do the permutation needed BEFORE calling ff_fft_calc().
static int process_get_band_noise(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int band)
static int config_input(AVFilterLink *inlink)
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
static void factor(double *array, int size)
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static double freq2bark(double x)
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
double noise_band_sample[15]
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function.If this function returns true
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
A filter pad used for either input or output.
static int query_formats(AVFilterContext *ctx)
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
A link between two filters.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
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...
void * priv
private data for use by the filter
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
double * prior_band_excit
FFTContext * av_fft_init(int nbits, int inverse)
Set up a complex FFT.
static int get_band_edge(AudioFFTDeNoiseContext *s, int band)
static void get_auto_noise_levels(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *levels)
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
double noise_band_auto_var[15]
Context for an Audio FIFO Buffer.
float last_noise_reduction
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
int channels
number of audio channels, only used for audio.
audio channel layout utility functions
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
static av_cold void uninit(AVFilterContext *ctx)
static int get_band_noise(AudioFFTDeNoiseContext *s, int band, double a, double b, double c)
float fmaxf(float, float)
static void read_custom_noise(AudioFFTDeNoiseContext *s, int ch)
static void set_band_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch)
static void preprocess(FFTComplex *in, int len)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static int output_frame(AVFilterLink *inlink)
static void postprocess(FFTComplex *in, int len)
double noise_band_avr[15]
int format
agreed upon media format
A list of supported channel layouts.
#define AV_LOG_INFO
Standard information.
char * av_strdup(const char *s)
Duplicate a string.
AVSampleFormat
Audio sample formats.
Used for passing data between threads.
static const AVFilterPad outputs[]
double fmax(double, double)
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
double noise_band_avi[15]
static double limit_gain(double a, double b)
Rational number (pair of numerator and denominator).
AVFILTER_DEFINE_CLASS(afftdn)
const char * name
Filter name.
static void finish_sample_noise(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *sample_noise)
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
static void solve(double *matrix, double *vector, int size)
#define flags(name, subs,...)
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
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
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
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()...
float last_residual_floor
static void process_frame(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, FFTComplex *fft_data, double *prior, double *prior_band_excit, int track_noise)
static int activate(AVFilterContext *ctx)
double fmin(double, double)
int channels
Number of channels.
avfilter_execute_func * execute
AVFilterContext * dst
dest filter
static void init_sample_noise(DeNoiseChannel *dnch)
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
Peek data from an AVAudioFifo.
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static int array[MAX_W *MAX_W]
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
uint8_t ** extended_data
pointers to the data planes/channels.
static void sample_noise_block(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, AVFrame *in, int ch)
void av_fft_calc(FFTContext *s, FFTComplex *z)
Do a complex FFT with the parameters defined in av_fft_init().
static int get_band_centre(AudioFFTDeNoiseContext *s, int band)
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_NOPTS_VALUE
Undefined timestamp value.
static void calculate_sfm(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int start, int end)