Attached.
From ad9def7fad58a75450176413564543a16965d165 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <one...@gmail.com>
Date: Sun, 13 Aug 2023 05:03:00 +0200
Subject: [PATCH 3/3] avfilter/af_asdr: use single structure for sums
Signed-off-by: Paul B Mahol <one...@gmail.com>
---
libavfilter/af_asdr.c | 44 +++++++++++++++++++++++--------------------
1 file changed, 24 insertions(+), 20 deletions(-)
diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
index 53069427bf..dbbb7e3419 100644
--- a/libavfilter/af_asdr.c
+++ b/libavfilter/af_asdr.c
@@ -27,13 +27,18 @@
#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;
- double *sum_u;
- double *sum_v;
- double *sum_uv;
+
+ ChanStats *chs;
AVFrame *cache[2];
@@ -52,6 +57,7 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int jobnr, int 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.; \
@@ -62,8 +68,8 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)\
sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
} \
\
- s->sum_uv[ch] += sum_uv; \
- s->sum_u[ch] += sum_u; \
+ chs->uv += sum_uv; \
+ chs->u += sum_u; \
} \
\
return 0; \
@@ -84,6 +90,7 @@ static int sisdr_##name(AVFilterContext *ctx, void *arg,int jobnr,int 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.; \
@@ -96,9 +103,9 @@ static int sisdr_##name(AVFilterContext *ctx, void *arg,int jobnr,int nb_jobs)\
sum_uv += us[n] * vs[n]; \
} \
\
- s->sum_uv[ch] += sum_uv; \
- s->sum_u[ch] += sum_u; \
- s->sum_v[ch] += sum_v; \
+ chs->uv += sum_uv; \
+ chs->u += sum_u; \
+ chs->v += sum_v; \
} \
\
return 0; \
@@ -119,6 +126,7 @@ static int psnr_##name(AVFilterContext *ctx, void *arg, int jobnr,int 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.; \
@@ -126,7 +134,7 @@ static int psnr_##name(AVFilterContext *ctx, void *arg, int jobnr,int nb_jobs)\
for (int n = 0; n < nb_samples; n++) \
sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
\
- s->sum_uv[ch] += sum_uv; \
+ chs->uv += sum_uv; \
} \
\
return 0; \
@@ -204,10 +212,8 @@ static int config_output(AVFilterLink *outlink)
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 || !s->sum_v)
+ s->chs = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->chs));
+ if (!s->chs)
return AVERROR(ENOMEM);
return 0;
@@ -219,17 +225,17 @@ 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]));
+ av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 20. * 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->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]);
+ double scale = s->chs[ch].uv / s->chs[ch].v;
+ double sisdr = s->chs[ch].u / 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->sum_uv[ch] > 0.0 ? 2.0 * log(s->max) - log(s->nb_samples / s->sum_uv[ch]) : INFINITY;
+ 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);
}
@@ -238,9 +244,7 @@ static av_cold void uninit(AVFilterContext *ctx)
av_frame_free(&s->cache[0]);
av_frame_free(&s->cache[1]);
- av_freep(&s->sum_u);
- av_freep(&s->sum_v);
- av_freep(&s->sum_uv);
+ av_freep(&s->chs);
}
static const AVFilterPad inputs[] = {
--
2.39.1
From dfb20b0f4d08428a43b38185776baf6819fc4336 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <one...@gmail.com>
Date: Sun, 13 Aug 2023 04:19:08 +0200
Subject: [PATCH 2/3] avfilter: add asisdr filter
Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 7 +++++
libavfilter/Makefile | 1 +
libavfilter/af_asdr.c | 64 +++++++++++++++++++++++++++++++++++++++-
libavfilter/allfilters.c | 1 +
4 files changed, 72 insertions(+), 1 deletion(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 1025917c63..159764bcb6 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3197,6 +3197,13 @@ 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.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 90c30e3388..ba07f4ab1e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -103,6 +103,7 @@ 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
diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
index b0401804f6..53069427bf 100644
--- a/libavfilter/af_asdr.c
+++ b/libavfilter/af_asdr.c
@@ -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),
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 949c5d4992..2cda06f251 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -89,6 +89,7 @@ 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;
--
2.39.1
From 8b457c83855ccc292a53be2bd716bf445d37d7e0 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <one...@gmail.com>
Date: Sun, 13 Aug 2023 02:57:57 +0200
Subject: [PATCH 1/3] avfilter: add apsnr filter
Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 7 +++++
libavfilter/Makefile | 1 +
libavfilter/af_asdr.c | 67 ++++++++++++++++++++++++++++++++++++++--
libavfilter/allfilters.c | 1 +
4 files changed, 73 insertions(+), 3 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index e041adc40f..1025917c63 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2892,6 +2892,13 @@ Default value is 8.
This filter supports the all above options as @ref{commands}.
+@section apsnr
+Measure Audio Peak Signal-to-Noise 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 apsyclip
Apply Psychoacoustic clipper to input audio stream.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index b26a85a7d2..90c30e3388 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -85,6 +85,7 @@ OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
+OBJS-$(CONFIG_APSNR_FILTER) += af_asdr.o
OBJS-$(CONFIG_APSYCLIP_FILTER) += af_apsyclip.o
OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
index 7d778b7f6b..b0401804f6 100644
--- a/libavfilter/af_asdr.c
+++ b/libavfilter/af_asdr.c
@@ -18,6 +18,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include <float.h>
+
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
@@ -27,6 +29,8 @@
typedef struct AudioSDRContext {
int channels;
+ uint64_t nb_samples;
+ double max;
double *sum_u;
double *sum_uv;
@@ -67,6 +71,34 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)\
SDR_FILTER(fltp, float)
SDR_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++) { \
+ 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]); \
+ \
+ s->sum_uv[ch] += sum_uv; \
+ } \
+ \
+ return 0; \
+}
+
+PSNR_FILTER(fltp, float)
+PSNR_FILTER(dblp, double)
+
static int activate(AVFilterContext *ctx)
{
AudioSDRContext *s = ctx->priv;
@@ -97,6 +129,7 @@ static int activate(AVFilterContext *ctx)
out = s->cache[0];
s->cache[0] = NULL;
+ s->nb_samples += available;
return ff_filter_frame(outlink, out);
}
@@ -126,7 +159,12 @@ static int config_output(AVFilterLink *outlink)
AudioSDRContext *s = ctx->priv;
s->channels = inlink->ch_layout.nb_channels;
- s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
+
+ if (!strcmp(ctx->filter->name, "asdr"))
+ s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_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_uv = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->sum_uv));
@@ -140,8 +178,16 @@ static av_cold void uninit(AVFilterContext *ctx)
{
AudioSDRContext *s = ctx->priv;
- 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]));
+ 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 {
+ 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;
+
+ 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]);
@@ -183,3 +229,18 @@ const AVFilter ff_af_asdr = {
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),
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 286e601799..949c5d4992 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -71,6 +71,7 @@ extern const AVFilter ff_af_apad;
extern const AVFilter ff_af_aperms;
extern const AVFilter ff_af_aphaser;
extern const AVFilter ff_af_aphaseshift;
+extern const AVFilter ff_af_apsnr;
extern const AVFilter ff_af_apsyclip;
extern const AVFilter ff_af_apulsator;
extern const AVFilter ff_af_arealtime;
--
2.39.1
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