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avfilter/af_afftdn: use better approach for gain smoothing

This commit is contained in:
Paul B Mahol 2022-04-23 22:18:26 +02:00
parent eae3b84189
commit fe85afbf8c
2 changed files with 30 additions and 12 deletions
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7
doc/filters.texi
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@ -1392,10 +1392,11 @@ Default value is @code{none}.
@end table
@item gain_smooth, gs
Set gain smooth factor, used to lowpass gains applied to each frequency bin.
Set gain smooth spatial radius, used to smooth gains applied to each frequency bin.
Useful to reduce random music noise artefacts.
Higher values reduce smoothing of gains.
Default value is @code{1.0}.
Higher values increases smoothing of gains.
Allowed range is from @code{0} to @code{50}.
Default value is @code{0}.
@end table
@subsection Commands

35
libavfilter/af_afftdn.c
View File

@ -71,6 +71,7 @@ typedef struct DeNoiseChannel {
double *band_amt;
double *band_excit;
double *gain;
double *smoothed_gain;
double *prior;
double *prior_band_excit;
double *clean_data;
@ -114,7 +115,7 @@ typedef struct AudioFFTDeNoiseContext {
int output_mode;
int noise_floor_link;
float ratio;
float gain_smooth;
int gain_smooth;
float band_multiplier;
float floor_offset;
@ -208,8 +209,8 @@ static const AVOption afftdn_options[] = {
{ "begin", "start", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_START}, 0, 0, AFR, "sample" },
{ "stop", "stop", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_STOP}, 0, 0, AFR, "sample" },
{ "end", "stop", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_STOP}, 0, 0, AFR, "sample" },
{ "gain_smooth", "set gain smooth factor",OFFSET(gain_smooth), AV_OPT_TYPE_FLOAT, {.dbl = 1.00}, 0.0001, 1, AFR },
{ "gs", "set gain smooth factor",OFFSET(gain_smooth), AV_OPT_TYPE_FLOAT, {.dbl = 1.00}, 0.0001, 1, AFR },
{ "gain_smooth", "set gain smooth radius",OFFSET(gain_smooth), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 50, AFR },
{ "gs", "set gain smooth radius",OFFSET(gain_smooth), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 50, AFR },
{ NULL }
};
@ -357,6 +358,7 @@ static void process_frame(AVFilterContext *ctx,
double *noisy_data = dnch->noisy_data;
double *band_excit = dnch->band_excit;
double *band_amt = dnch->band_amt;
double *smoothed_gain = dnch->smoothed_gain;
double *gain = dnch->gain;
for (int i = 0; i < s->bin_count; i++) {
@ -424,16 +426,28 @@ static void process_frame(AVFilterContext *ctx,
}
}
{
const double f = s->gain_smooth;
const double F = 1. - f;
memcpy(smoothed_gain, gain, s->bin_count * sizeof(*smoothed_gain));
if (s->gain_smooth > 0) {
const int r = s->gain_smooth;
for (int i = 1; i < s->bin_count; i++)
gain[i] = gain[i-1] * F + f * gain[i];
for (int i = r; i < s->bin_count - r; i++) {
const double gc = gain[i];
double num = 0., den = 0.;
for (int j = -r; j <= r; j++) {
const double g = gain[i + j];
const double d = 1. - fabs(g - gc);
num += g * d;
den += d;
}
smoothed_gain[i] = num / den;
}
}
for (int i = 0; i < s->bin_count; i++) {
const double new_gain = gain[i];
const double new_gain = smoothed_gain[i];
fft_data[i].re *= new_gain;
fft_data[i].im *= new_gain;
@ -685,6 +699,7 @@ static int config_input(AVFilterLink *inlink)
dnch->band_amt = av_calloc(s->number_of_bands, sizeof(*dnch->band_amt));
dnch->band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->band_excit));
dnch->gain = av_calloc(s->bin_count, sizeof(*dnch->gain));
dnch->smoothed_gain = av_calloc(s->bin_count, sizeof(*dnch->smoothed_gain));
dnch->prior = av_calloc(s->bin_count, sizeof(*dnch->prior));
dnch->prior_band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->prior_band_excit));
dnch->clean_data = av_calloc(s->bin_count, sizeof(*dnch->clean_data));
@ -708,6 +723,7 @@ static int config_input(AVFilterLink *inlink)
!dnch->band_amt ||
!dnch->band_excit ||
!dnch->gain ||
!dnch->smoothed_gain ||
!dnch->prior ||
!dnch->prior_band_excit ||
!dnch->clean_data ||
@ -1161,6 +1177,7 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&dnch->band_amt);
av_freep(&dnch->band_excit);
av_freep(&dnch->gain);
av_freep(&dnch->smoothed_gain);
av_freep(&dnch->prior);
av_freep(&dnch->prior_band_excit);
av_freep(&dnch->clean_data);