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avfilter/af_arls: add double sample format support

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This commit is contained in:
Paul B Mahol 2023-11-27 20:13:31 +01:00
parent 42e45ea8ff
commit 3bca828d39
3 changed files with 221 additions and 107 deletions
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14
doc/filters.texi
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@ -3041,6 +3041,20 @@ Pass error signal estimated samples.
Default value is @var{o}.
@end table
@item precision
Set which precision to use when processing samples.
@table @option
@item auto
Auto pick internal sample format depending on other filters.
@item float
Always use single-floating point precision sample format.
@item double
Always use double-floating point precision sample format.
@end table
@end table
@section arnndn

156
libavfilter/af_arls.c
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@ -24,6 +24,7 @@
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "filters.h"
#include "internal.h"
@ -43,6 +44,7 @@ typedef struct AudioRLSContext {
float lambda;
float delta;
int output_mode;
int precision;
int kernel_size;
AVFrame *offset;
@ -54,6 +56,8 @@ typedef struct AudioRLSContext {
AVFrame *frame[2];
int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
AVFloatDSPContext *fdsp;
} AudioRLSContext;
@ -71,117 +75,32 @@ static const AVOption arls_options[] = {
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" },
{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" },
{ "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, "mode" },
{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "precision" },
{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "precision" },
{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "precision" },
{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "precision" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(arls);
static float fir_sample(AudioRLSContext *s, float sample, float *delay,
float *coeffs, float *tmp, int *offset)
{
const int order = s->order;
float output;
delay[*offset] = sample;
memcpy(tmp, coeffs + order - *offset, order * sizeof(float));
output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
if (--(*offset) < 0)
*offset = order - 1;
return output;
}
static float process_sample(AudioRLSContext *s, float input, float desired, int ch)
{
float *coeffs = (float *)s->coeffs->extended_data[ch];
float *delay = (float *)s->delay->extended_data[ch];
float *gains = (float *)s->gains->extended_data[ch];
float *tmp = (float *)s->tmp->extended_data[ch];
float *u = (float *)s->u->extended_data[ch];
float *p = (float *)s->p->extended_data[ch];
float *dp = (float *)s->dp->extended_data[ch];
int *offsetp = (int *)s->offset->extended_data[ch];
const int kernel_size = s->kernel_size;
const int order = s->order;
const float lambda = s->lambda;
int offset = *offsetp;
float g = lambda;
float output, e;
delay[offset + order] = input;
output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
e = desired - output;
for (int i = 0, pos = offset; i < order; i++, pos++) {
const int ikernel_size = i * kernel_size;
u[i] = 0.f;
for (int k = 0, pos = offset; k < order; k++, pos++)
u[i] += p[ikernel_size + k] * delay[pos];
g += u[i] * delay[pos];
}
g = 1.f / g;
for (int i = 0; i < order; i++) {
const int ikernel_size = i * kernel_size;
gains[i] = u[i] * g;
coeffs[i] = coeffs[order + i] = coeffs[i] + gains[i] * e;
tmp[i] = 0.f;
for (int k = 0, pos = offset; k < order; k++, pos++)
tmp[i] += p[ikernel_size + k] * delay[pos];
}
for (int i = 0; i < order; i++) {
const int ikernel_size = i * kernel_size;
for (int k = 0; k < order; k++)
dp[ikernel_size + k] = gains[i] * tmp[k];
}
for (int i = 0; i < order; i++) {
const int ikernel_size = i * kernel_size;
for (int k = 0; k < order; k++)
p[ikernel_size + k] = (p[ikernel_size + k] - (dp[ikernel_size + k] + dp[kernel_size * k + i]) * 0.5f) * lambda;
}
switch (s->output_mode) {
case IN_MODE: output = input; break;
case DESIRED_MODE: output = desired; break;
case OUT_MODE: output = desired - output; break;
case NOISE_MODE: output = input - output; break;
case ERROR_MODE: break;
}
return output;
}
static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
static int query_formats(AVFilterContext *ctx)
{
AudioRLSContext *s = ctx->priv;
AVFrame *out = arg;
const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
static const enum AVSampleFormat sample_fmts[3][3] = {
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
};
int ret;
for (int c = start; c < end; c++) {
const float *input = (const float *)s->frame[0]->extended_data[c];
const float *desired = (const float *)s->frame[1]->extended_data[c];
float *output = (float *)out->extended_data[c];
if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
return ret;
for (int n = 0; n < out->nb_samples; n++) {
output[n] = process_sample(s, input[n], desired[n], c);
if (ctx->is_disabled)
output[n] = input[n];
}
}
if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
return ret;
return 0;
return ff_set_common_all_samplerates(ctx);
}
static int activate(AVFilterContext *ctx)
@ -216,7 +135,7 @@ static int activate(AVFilterContext *ctx)
return AVERROR(ENOMEM);
}
ff_filter_execute(ctx, process_channels, out, NULL,
ff_filter_execute(ctx, s->filter_channels, out, NULL,
FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
out->pts = s->frame[0]->pts;
@ -249,6 +168,13 @@ static int activate(AVFilterContext *ctx)
return 0;
}
#define DEPTH 32
#include "arls_template.c"
#undef DEPTH
#define DEPTH 64
#include "arls_template.c"
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
@ -283,11 +209,27 @@ static int config_output(AVFilterLink *outlink)
dst[0] = s->kernel_size - 1;
}
for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) {
float *dst = (float *)s->p->extended_data[ch];
switch (outlink->format) {
case AV_SAMPLE_FMT_DBLP:
for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) {
double *dst = (double *)s->p->extended_data[ch];
for (int i = 0; i < s->kernel_size; i++)
dst[i * s->kernel_size + i] = s->delta;
for (int i = 0; i < s->kernel_size; i++)
dst[i * s->kernel_size + i] = s->delta;
}
s->filter_channels = filter_channels_double;
break;
case AV_SAMPLE_FMT_FLTP:
for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) {
float *dst = (float *)s->p->extended_data[ch];
for (int i = 0; i < s->kernel_size; i++)
dst[i * s->kernel_size + i] = s->delta;
}
s->filter_channels = filter_channels_float;
break;
}
return 0;
@ -348,7 +290,7 @@ const AVFilter ff_af_arls = {
.activate = activate,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,

158
libavfilter/arls_template.c Normal file
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@ -0,0 +1,158 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#undef ONE
#undef ftype
#undef SAMPLE_FORMAT
#if DEPTH == 32
#define SAMPLE_FORMAT float
#define ftype float
#define ONE 1.f
#else
#define SAMPLE_FORMAT double
#define ftype double
#define ONE 1.0
#endif
#define fn3(a,b) a##_##b
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
#if DEPTH == 64
static double scalarproduct_double(const double *v1, const double *v2, int len)
{
double p = 0.0;
for (int i = 0; i < len; i++)
p += v1[i] * v2[i];
return p;
}
#endif
static ftype fn(fir_sample)(AudioRLSContext *s, ftype sample, ftype *delay,
ftype *coeffs, ftype *tmp, int *offset)
{
const int order = s->order;
ftype output;
delay[*offset] = sample;
memcpy(tmp, coeffs + order - *offset, order * sizeof(ftype));
#if DEPTH == 32
output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
#else
output = scalarproduct_double(delay, tmp, s->kernel_size);
#endif
if (--(*offset) < 0)
*offset = order - 1;
return output;
}
static ftype fn(process_sample)(AudioRLSContext *s, ftype input, ftype desired, int ch)
{
ftype *coeffs = (ftype *)s->coeffs->extended_data[ch];
ftype *delay = (ftype *)s->delay->extended_data[ch];
ftype *gains = (ftype *)s->gains->extended_data[ch];
ftype *tmp = (ftype *)s->tmp->extended_data[ch];
ftype *u = (ftype *)s->u->extended_data[ch];
ftype *p = (ftype *)s->p->extended_data[ch];
ftype *dp = (ftype *)s->dp->extended_data[ch];
int *offsetp = (int *)s->offset->extended_data[ch];
const int kernel_size = s->kernel_size;
const int order = s->order;
const ftype lambda = s->lambda;
int offset = *offsetp;
ftype g = lambda;
ftype output, e;
delay[offset + order] = input;
output = fn(fir_sample)(s, input, delay, coeffs, tmp, offsetp);
e = desired - output;
for (int i = 0, pos = offset; i < order; i++, pos++) {
const int ikernel_size = i * kernel_size;
u[i] = 0.f;
for (int k = 0, pos = offset; k < order; k++, pos++)
u[i] += p[ikernel_size + k] * delay[pos];
g += u[i] * delay[pos];
}
g = 1.f / g;
for (int i = 0; i < order; i++) {
const int ikernel_size = i * kernel_size;
gains[i] = u[i] * g;
coeffs[i] = coeffs[order + i] = coeffs[i] + gains[i] * e;
tmp[i] = 0.f;
for (int k = 0, pos = offset; k < order; k++, pos++)
tmp[i] += p[ikernel_size + k] * delay[pos];
}
for (int i = 0; i < order; i++) {
const int ikernel_size = i * kernel_size;
for (int k = 0; k < order; k++)
dp[ikernel_size + k] = gains[i] * tmp[k];
}
for (int i = 0; i < order; i++) {
const int ikernel_size = i * kernel_size;
for (int k = 0; k < order; k++)
p[ikernel_size + k] = (p[ikernel_size + k] - (dp[ikernel_size + k] + dp[kernel_size * k + i]) * 0.5f) * lambda;
}
switch (s->output_mode) {
case IN_MODE: output = input; break;
case DESIRED_MODE: output = desired; break;
case OUT_MODE: output = desired - output; break;
case NOISE_MODE: output = input - output; break;
case ERROR_MODE: break;
}
return output;
}
static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioRLSContext *s = ctx->priv;
AVFrame *out = arg;
const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
for (int c = start; c < end; c++) {
const ftype *input = (const ftype *)s->frame[0]->extended_data[c];
const ftype *desired = (const ftype *)s->frame[1]->extended_data[c];
ftype *output = (ftype *)out->extended_data[c];
for (int n = 0; n < out->nb_samples; n++) {
output[n] = fn(process_sample)(s, input[n], desired[n], c);
if (ctx->is_disabled)
output[n] = input[n];
}
}
return 0;
}