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/*
* Copyright (c) 2021 Paul B Mahol
*
* 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
*/
#include <float.h>
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/mem.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
typedef struct ChanStats {
double u;
double v;
double uv;
} ChanStats;
typedef struct AudioSDRContext {
int channels;
uint64_t nb_samples;
double max;
ChanStats *chs;
AVFrame *cache[2];
int (*filter)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
} AudioSDRContext;
#define SDR_FILTER(name, type) \
static int sdr_##name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)\
{ \
AudioSDRContext *s = ctx->priv; \
AVFrame *u = s->cache[0]; \
AVFrame *v = s->cache[1]; \
const int channels = u->ch_layout.nb_channels; \
const int start = (channels * jobnr) / nb_jobs; \
const int end = (channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = u->nb_samples; \
\
for (int ch = start; ch < end; ch++) { \
ChanStats *chs = &s->chs[ch]; \
const type *const us = (type *)u->extended_data[ch]; \
const type *const vs = (type *)v->extended_data[ch]; \
double sum_uv = 0.; \
double sum_u = 0.; \
\
for (int n = 0; n < nb_samples; n++) { \
sum_u += us[n] * us[n]; \
sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
} \
\
chs->uv += sum_uv; \
chs->u += sum_u; \
} \
\
return 0; \
}
SDR_FILTER(fltp, float)
SDR_FILTER(dblp, double)
#define SISDR_FILTER(name, type) \
static int sisdr_##name(AVFilterContext *ctx, void *arg,int jobnr,int nb_jobs)\
{ \
AudioSDRContext *s = ctx->priv; \
AVFrame *u = s->cache[0]; \
AVFrame *v = s->cache[1]; \
const int channels = u->ch_layout.nb_channels; \
const int start = (channels * jobnr) / nb_jobs; \
const int end = (channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = u->nb_samples; \
\
for (int ch = start; ch < end; ch++) { \
ChanStats *chs = &s->chs[ch]; \
const type *const us = (type *)u->extended_data[ch]; \
const type *const vs = (type *)v->extended_data[ch]; \
double sum_uv = 0.; \
double sum_u = 0.; \
double sum_v = 0.; \
\
for (int n = 0; n < nb_samples; n++) { \
sum_u += us[n] * us[n]; \
sum_v += vs[n] * vs[n]; \
sum_uv += us[n] * vs[n]; \
} \
\
chs->uv += sum_uv; \
chs->u += sum_u; \
chs->v += sum_v; \
} \
\
return 0; \
}
SISDR_FILTER(fltp, float)
SISDR_FILTER(dblp, double)
#define PSNR_FILTER(name, type) \
static int psnr_##name(AVFilterContext *ctx, void *arg, int jobnr,int nb_jobs)\
{ \
AudioSDRContext *s = ctx->priv; \
AVFrame *u = s->cache[0]; \
AVFrame *v = s->cache[1]; \
const int channels = u->ch_layout.nb_channels; \
const int start = (channels * jobnr) / nb_jobs; \
const int end = (channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = u->nb_samples; \
\
for (int ch = start; ch < end; ch++) { \
ChanStats *chs = &s->chs[ch]; \
const type *const us = (type *)u->extended_data[ch]; \
const type *const vs = (type *)v->extended_data[ch]; \
double sum_uv = 0.; \
\
for (int n = 0; n < nb_samples; n++) \
sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
\
chs->uv += sum_uv; \
} \
\
return 0; \
}
PSNR_FILTER(fltp, float)
PSNR_FILTER(dblp, double)
static int activate(AVFilterContext *ctx)
{
AudioSDRContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret, status, available;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
available = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), ff_inlink_queued_samples(ctx->inputs[1]));
if (available > 0) {
AVFrame *out;
for (int i = 0; i < 2; i++) {
ret = ff_inlink_consume_samples(ctx->inputs[i], available, available, &s->cache[i]);
if (ret < 0) {
av_frame_free(&s->cache[0]);
av_frame_free(&s->cache[1]);
return ret;
}
}
if (!ctx->is_disabled)
ff_filter_execute(ctx, s->filter, NULL, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
av_frame_free(&s->cache[1]);
out = s->cache[0];
s->cache[0] = NULL;
s->nb_samples += available;
return ff_filter_frame(outlink, out);
}
for (int i = 0; i < 2; i++) {
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
ff_outlink_set_status(outlink, status, pts);
return 0;
}
}
if (ff_outlink_frame_wanted(outlink)) {
for (int i = 0; i < 2; i++) {
if (s->cache[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0)
continue;
ff_inlink_request_frame(ctx->inputs[i]);
return 0;
}
}
return FFERROR_NOT_READY;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AudioSDRContext *s = ctx->priv;
s->channels = inlink->ch_layout.nb_channels;
if (!strcmp(ctx->filter->name, "asdr"))
s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
else if (!strcmp(ctx->filter->name, "asisdr"))
s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sisdr_fltp : sisdr_dblp;
else
s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? psnr_fltp : psnr_dblp;
s->max = inlink->format == AV_SAMPLE_FMT_FLTP ? FLT_MAX : DBL_MAX;
s->chs = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->chs));
if (!s->chs)
return AVERROR(ENOMEM);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioSDRContext *s = ctx->priv;
if (!strcmp(ctx->filter->name, "asdr")) {
for (int ch = 0; ch < s->channels; ch++)
av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 10. * log10(s->chs[ch].u / s->chs[ch].uv));
} else if (!strcmp(ctx->filter->name, "asisdr")) {
for (int ch = 0; ch < s->channels; ch++) {
double scale = s->chs[ch].uv / s->chs[ch].v;
double sisdr = scale * scale * s->chs[ch].v / fmax(0., s->chs[ch].u + scale*scale*s->chs[ch].v - 2.0*scale*s->chs[ch].uv);
av_log(ctx, AV_LOG_INFO, "SI-SDR ch%d: %g dB\n", ch, 10. * log10(sisdr));
}
} else {
for (int ch = 0; ch < s->channels; ch++) {
double psnr = s->chs[ch].uv > 0.0 ? 2.0 * log(s->max) - log(s->nb_samples / s->chs[ch].uv) : INFINITY;
av_log(ctx, AV_LOG_INFO, "PSNR ch%d: %g dB\n", ch, psnr);
}
}
av_frame_free(&s->cache[0]);
av_frame_free(&s->cache[1]);
av_freep(&s->chs);
}
static const AVFilterPad inputs[] = {
{
.name = "input0",
.type = AVMEDIA_TYPE_AUDIO,
},
{
.name = "input1",
.type = AVMEDIA_TYPE_AUDIO,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_asdr = {
.name = "asdr",
.description = NULL_IF_CONFIG_SMALL("Measure Audio Signal-to-Distortion Ratio."),
.priv_size = sizeof(AudioSDRContext),
.activate = activate,
.uninit = uninit,
.flags = AVFILTER_FLAG_METADATA_ONLY |
AVFILTER_FLAG_SLICE_THREADS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP),
};
const AVFilter ff_af_apsnr = {
.name = "apsnr",
.description = NULL_IF_CONFIG_SMALL("Measure Audio Peak Signal-to-Noise Ratio."),
.priv_size = sizeof(AudioSDRContext),
.activate = activate,
.uninit = uninit,
.flags = AVFILTER_FLAG_METADATA_ONLY |
AVFILTER_FLAG_SLICE_THREADS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP),
};
const AVFilter ff_af_asisdr = {
.name = "asisdr",
.description = NULL_IF_CONFIG_SMALL("Measure Audio Scale-Invariant Signal-to-Distortion Ratio."),
.priv_size = sizeof(AudioSDRContext),
.activate = activate,
.uninit = uninit,
.flags = AVFILTER_FLAG_METADATA_ONLY |
AVFILTER_FLAG_SLICE_THREADS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP),
};
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