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avfilter/af_afftdn: add double sample format support

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This commit is contained in:
Paul B Mahol 2022-09-18 22:30:48 +02:00
parent 1af0051977
commit 1f9f0dc6ee
1 changed files with 170 additions and 48 deletions
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218
libavfilter/af_afftdn.c
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@ -81,8 +81,8 @@ typedef struct DeNoiseChannel {
double *abs_var;
double *rel_var;
double *min_abs_var;
float *fft_in;
AVComplexFloat *fft_out;
void *fft_in;
void *fft_out;
AVTXContext *fft, *ifft;
av_tx_fn tx_fn, itx_fn;
@ -105,6 +105,9 @@ typedef struct DeNoiseChannel {
typedef struct AudioFFTDeNoiseContext {
const AVClass *class;
int format;
size_t sample_size;
float noise_reduction;
float noise_floor;
int noise_type;
@ -347,7 +350,6 @@ static double floor_offset(const double *S, int size, double mean)
static void process_frame(AVFilterContext *ctx,
AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch,
AVComplexFloat *fft_data,
double *prior, double *prior_band_excit, int track_noise)
{
AVFilterLink *outlink = ctx->outputs[0];
@ -359,12 +361,22 @@ static void process_frame(AVFilterContext *ctx,
double *band_excit = dnch->band_excit;
double *band_amt = dnch->band_amt;
double *smoothed_gain = dnch->smoothed_gain;
AVComplexDouble *fft_data_dbl = dnch->fft_out;
AVComplexFloat *fft_data_flt = dnch->fft_out;
double *gain = dnch->gain;
for (int i = 0; i < s->bin_count; i++) {
double sqr_new_gain, new_gain, power, mag, mag_abs_var, new_mag_abs_var;
noisy_data[i] = mag = hypot(fft_data[i].re, fft_data[i].im);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
noisy_data[i] = mag = hypot(fft_data_flt[i].re, fft_data_flt[i].im);
break;
case AV_SAMPLE_FMT_DBLP:
noisy_data[i] = mag = hypot(fft_data_dbl[i].re, fft_data_dbl[i].im);
break;
}
power = mag * mag;
mag_abs_var = power / abs_var[i];
new_mag_abs_var = ratio * prior[i] + rratio * fmax(mag_abs_var - 1.0, 0.0);
@ -446,11 +458,23 @@ static void process_frame(AVFilterContext *ctx,
}
}
for (int i = 0; i < s->bin_count; i++) {
const double new_gain = smoothed_gain[i];
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int i = 0; i < s->bin_count; i++) {
const float new_gain = smoothed_gain[i];
fft_data[i].re *= new_gain;
fft_data[i].im *= new_gain;
fft_data_flt[i].re *= new_gain;
fft_data_flt[i].im *= new_gain;
}
break;
case AV_SAMPLE_FMT_DBLP:
for (int i = 0; i < s->bin_count; i++) {
const double new_gain = smoothed_gain[i];
fft_data_dbl[i].re *= new_gain;
fft_data_dbl[i].im *= new_gain;
}
break;
}
}
@ -603,8 +627,29 @@ static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioFFTDeNoiseContext *s = ctx->priv;
size_t complex_sample_size;
double wscale, sar, sum, sdiv;
int i, j, k, m, n, ret;
int i, j, k, m, n, ret, tx_type;
double dscale = 1.;
float fscale = 1.f;
void *scale;
s->format = inlink->format;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
s->sample_size = sizeof(float);
complex_sample_size = sizeof(AVComplexFloat);
tx_type = AV_TX_FLOAT_RDFT;
scale = &fscale;
break;
case AV_SAMPLE_FMT_DBLP:
s->sample_size = sizeof(double);
complex_sample_size = sizeof(AVComplexDouble);
tx_type = AV_TX_DOUBLE_RDFT;
scale = &dscale;
break;
}
s->dnch = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->dnch));
if (!s->dnch)
@ -671,7 +716,6 @@ static int config_input(AVFilterLink *inlink)
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
float scale = 1.f;
switch (s->noise_type) {
case WHITE_NOISE:
@ -708,12 +752,12 @@ static int config_input(AVFilterLink *inlink)
dnch->abs_var = av_calloc(s->bin_count, sizeof(*dnch->abs_var));
dnch->rel_var = av_calloc(s->bin_count, sizeof(*dnch->rel_var));
dnch->min_abs_var = av_calloc(s->bin_count, sizeof(*dnch->min_abs_var));
dnch->fft_in = av_calloc(s->fft_length2, sizeof(*dnch->fft_in));
dnch->fft_out = av_calloc(s->fft_length2 + 1, sizeof(*dnch->fft_out));
ret = av_tx_init(&dnch->fft, &dnch->tx_fn, AV_TX_FLOAT_RDFT, 0, s->fft_length2, &scale, 0);
dnch->fft_in = av_calloc(s->fft_length2, s->sample_size);
dnch->fft_out = av_calloc(s->fft_length2 + 1, complex_sample_size);
ret = av_tx_init(&dnch->fft, &dnch->tx_fn, tx_type, 0, s->fft_length2, scale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&dnch->ifft, &dnch->itx_fn, AV_TX_FLOAT_RDFT, 1, s->fft_length2, &scale, 0);
ret = av_tx_init(&dnch->ifft, &dnch->itx_fn, tx_type, 1, s->fft_length2, scale, 0);
if (ret < 0)
return ret;
dnch->spread_function = av_calloc(s->number_of_bands * s->number_of_bands,
@ -861,17 +905,33 @@ static void sample_noise_block(AudioFFTDeNoiseContext *s,
DeNoiseChannel *dnch,
AVFrame *in, int ch)
{
float *src = (float *)in->extended_data[ch];
double *src_dbl = (double *)in->extended_data[ch];
float *src_flt = (float *)in->extended_data[ch];
double mag2, var = 0.0, avr = 0.0, avi = 0.0;
AVComplexDouble *fft_out_dbl = dnch->fft_out;
AVComplexFloat *fft_out_flt = dnch->fft_out;
double *fft_in_dbl = dnch->fft_in;
float *fft_in_flt = dnch->fft_in;
int edge, j, k, n, edgemax;
for (int i = 0; i < s->window_length; i++)
dnch->fft_in[i] = s->window[i] * src[i] * (1LL << 23);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int i = 0; i < s->window_length; i++)
fft_in_flt[i] = s->window[i] * src_flt[i] * (1LL << 23);
for (int i = s->window_length; i < s->fft_length2; i++)
dnch->fft_in[i] = 0.0;
for (int i = s->window_length; i < s->fft_length2; i++)
fft_in_flt[i] = 0.f;
break;
case AV_SAMPLE_FMT_DBLP:
for (int i = 0; i < s->window_length; i++)
fft_in_dbl[i] = s->window[i] * src_dbl[i] * (1LL << 23);
dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, sizeof(float));
for (int i = s->window_length; i < s->fft_length2; i++)
fft_in_dbl[i] = 0.;
break;
}
dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, sizeof(s->sample_size));
edge = s->noise_band_edge[0];
j = edge;
@ -896,10 +956,21 @@ static void sample_noise_block(AudioFFTDeNoiseContext *s,
avr = 0.0;
avi = 0.0;
}
avr += dnch->fft_out[n].re;
avi += dnch->fft_out[n].im;
mag2 = dnch->fft_out[n].re * dnch->fft_out[n].re +
dnch->fft_out[n].im * dnch->fft_out[n].im;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
avr += fft_out_flt[n].re;
avi += fft_out_flt[n].im;
mag2 = fft_out_flt[n].re * fft_out_flt[n].re +
fft_out_flt[n].im * fft_out_flt[n].im;
break;
case AV_SAMPLE_FMT_DBLP:
avr += fft_out_dbl[n].re;
avi += fft_out_dbl[n].im;
mag2 = fft_out_dbl[n].re * fft_out_dbl[n].re +
fft_out_dbl[n].im * fft_out_dbl[n].im;
break;
}
mag2 = fmax(mag2, s->sample_floor);
@ -980,27 +1051,48 @@ static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_job
for (int ch = start; ch < end; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
const float *src = (const float *)in->extended_data[ch];
const double *src_dbl = (const double *)in->extended_data[ch];
const float *src_flt = (const float *)in->extended_data[ch];
double *dst = dnch->out_samples;
float *fft_in = dnch->fft_in;
double *fft_in_dbl = dnch->fft_in;
float *fft_in_flt = dnch->fft_in;
for (int m = 0; m < window_length; m++)
fft_in[m] = window[m] * src[m] * (1LL << 23);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < window_length; m++)
fft_in_flt[m] = window[m] * src_flt[m] * (1LL << 23);
for (int m = window_length; m < s->fft_length2; m++)
fft_in[m] = 0;
for (int m = window_length; m < s->fft_length2; m++)
fft_in_flt[m] = 0.f;
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < window_length; m++)
fft_in_dbl[m] = window[m] * src_dbl[m] * (1LL << 23);
dnch->tx_fn(dnch->fft, dnch->fft_out, fft_in, sizeof(float));
for (int m = window_length; m < s->fft_length2; m++)
fft_in_dbl[m] = 0.;
break;
}
process_frame(ctx, s, dnch, dnch->fft_out,
dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, sizeof(s->sample_size));
process_frame(ctx, s, dnch,
dnch->prior,
dnch->prior_band_excit,
s->track_noise);
dnch->itx_fn(dnch->ifft, fft_in, dnch->fft_out, sizeof(float));
dnch->itx_fn(dnch->ifft, dnch->fft_in, dnch->fft_out, sizeof(s->sample_size));
for (int m = 0; m < window_length; m++)
dst[m] += s->window[m] * fft_in[m] / (1LL << 23);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < window_length; m++)
dst[m] += s->window[m] * fft_in_flt[m] / (1LL << 23);
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < window_length; m++)
dst[m] += s->window[m] * fft_in_dbl[m] / (1LL << 23);
break;
}
}
return 0;
@ -1016,11 +1108,14 @@ static int output_frame(AVFilterLink *inlink, AVFrame *in)
AVFrame *out;
for (int ch = 0; ch < s->channels; ch++) {
float *src = (float *)s->winframe->extended_data[ch];
uint8_t *src = (uint8_t *)s->winframe->extended_data[ch];
memmove(src, &src[s->sample_advance], offset * sizeof(float));
memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
memset(&src[offset + in->nb_samples], 0, (s->sample_advance - in->nb_samples) * sizeof(float));
memmove(src, src + s->sample_advance * s->sample_size,
offset * s->sample_size);
memcpy(src + offset * s->sample_size, in->extended_data[ch],
in->nb_samples * s->sample_size);
memset(src + s->sample_size * (offset + in->nb_samples), 0,
(s->sample_advance - in->nb_samples) * s->sample_size);
}
if (s->track_noise) {
@ -1107,21 +1202,47 @@ static int output_frame(AVFilterLink *inlink, AVFrame *in)
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
double *src = dnch->out_samples;
const float *orig = (const float *)s->winframe->extended_data[ch];
float *dst = (float *)out->extended_data[ch];
const double *orig_dbl = (const double *)s->winframe->extended_data[ch];
const float *orig_flt = (const float *)s->winframe->extended_data[ch];
double *dst_dbl = (double *)out->extended_data[ch];
float *dst_flt = (float *)out->extended_data[ch];
switch (output_mode) {
case IN_MODE:
for (int m = 0; m < out->nb_samples; m++)
dst[m] = orig[m];
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < out->nb_samples; m++)
dst_flt[m] = orig_flt[m];
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < out->nb_samples; m++)
dst_dbl[m] = orig_dbl[m];
break;
}
break;
case OUT_MODE:
for (int m = 0; m < out->nb_samples; m++)
dst[m] = src[m];
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < out->nb_samples; m++)
dst_flt[m] = src[m];
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < out->nb_samples; m++)
dst_dbl[m] = src[m];
break;
}
break;
case NOISE_MODE:
for (int m = 0; m < out->nb_samples; m++)
dst[m] = orig[m] - src[m];
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int m = 0; m < out->nb_samples; m++)
dst_flt[m] = orig_flt[m] - src[m];
break;
case AV_SAMPLE_FMT_DBLP:
for (int m = 0; m < out->nb_samples; m++)
dst_dbl[m] = orig_dbl[m] - src[m];
break;
}
break;
default:
if (in != out)
@ -1129,6 +1250,7 @@ static int output_frame(AVFilterLink *inlink, AVFrame *in)
av_frame_free(&out);
return AVERROR_BUG;
}
memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src));
memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
}
@ -1251,7 +1373,7 @@ const AVFilter ff_af_afftdn = {
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,