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avfilter: add asisdr filter

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This commit is contained in:
Paul B Mahol 2023-08-13 04:19:08 +02:00
parent 951def850a
commit e41d52216c
4 changed files with 72 additions and 1 deletions
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7
doc/filters.texi
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@ -3141,6 +3141,13 @@ audio, the data is treated as if all the planes were concatenated.
A list of Adler-32 checksums for each data plane.
@end table
@section asisdr
Measure Audio Scaled-Invariant Signal-to-Distortion Ratio.
This filter takes two audio streams for input, and outputs first
audio stream.
Results are in dB per channel at end of either input.
@section asoftclip
Apply audio soft clipping.

1
libavfilter/Makefile
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@ -102,6 +102,7 @@ OBJS-$(CONFIG_ASETRATE_FILTER) += af_asetrate.o
OBJS-$(CONFIG_ASETTB_FILTER) += settb.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASIDEDATA_FILTER) += f_sidedata.o
OBJS-$(CONFIG_ASISDR_FILTER) += af_asdr.o
OBJS-$(CONFIG_ASOFTCLIP_FILTER) += af_asoftclip.o
OBJS-$(CONFIG_ASPECTRALSTATS_FILTER) += af_aspectralstats.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o

64
libavfilter/af_asdr.c
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@ -32,6 +32,7 @@ typedef struct AudioSDRContext {
uint64_t nb_samples;
double max;
double *sum_u;
double *sum_v;
double *sum_uv;
AVFrame *cache[2];
@ -71,6 +72,41 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)\
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++) { \
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]; \
} \
\
s->sum_uv[ch] += sum_uv; \
s->sum_u[ch] += sum_u; \
s->sum_v[ch] += 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)\
{ \
@ -162,13 +198,16 @@ static int config_output(AVFilterLink *outlink)
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->sum_u = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->sum_u));
s->sum_v = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->sum_v));
s->sum_uv = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->sum_uv));
if (!s->sum_u || !s->sum_uv)
if (!s->sum_u || !s->sum_uv || !s->sum_v)
return AVERROR(ENOMEM);
return 0;
@ -181,6 +220,13 @@ static av_cold void uninit(AVFilterContext *ctx)
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, 20. * log10(s->sum_u[ch] / s->sum_uv[ch]));
} else if (!strcmp(ctx->filter->name, "asisdr")) {
for (int ch = 0; ch < s->channels; ch++) {
double scale = s->sum_uv[ch] / s->sum_v[ch];
double sisdr = s->sum_u[ch] / (s->sum_u[ch] + scale*scale*s->sum_v[ch] - 2.0*scale*s->sum_uv[ch]);
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->sum_uv[ch] > 0.0 ? 2.0 * log(s->max) - log(s->nb_samples / s->sum_uv[ch]) : INFINITY;
@ -193,6 +239,7 @@ static av_cold void uninit(AVFilterContext *ctx)
av_frame_free(&s->cache[1]);
av_freep(&s->sum_u);
av_freep(&s->sum_v);
av_freep(&s->sum_uv);
}
@ -244,3 +291,18 @@ const AVFilter ff_af_apsnr = {
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),
};

1
libavfilter/allfilters.c
View File

@ -88,6 +88,7 @@ extern const AVFilter ff_af_asetrate;
extern const AVFilter ff_af_asettb;
extern const AVFilter ff_af_ashowinfo;
extern const AVFilter ff_af_asidedata;
extern const AVFilter ff_af_asisdr;
extern const AVFilter ff_af_asoftclip;
extern const AVFilter ff_af_aspectralstats;
extern const AVFilter ff_af_asplit;