[FFmpeg-cvslog] avfilter: add asisdr filter
Paul B Mahol
git at videolan.org
Mon Aug 14 12:35:08 EEST 2023
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun Aug 13 04:19:08 2023 +0200| [e41d52216cfe3537d7eadca863fb25838edd18c6] | committer: Paul B Mahol
avfilter: add asisdr filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e41d52216cfe3537d7eadca863fb25838edd18c6
---
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 5d1f10f95d..cac1ee4381 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -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
+ at 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 9cd1407250..2fe0033b21 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
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 f6017c41c9..d4184d6e80 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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;
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