From dfb20b0f4d08428a43b38185776baf6819fc4336 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Sun, 13 Aug 2023 04:19:08 +0200 Subject: [PATCH 2/3] avfilter: add asisdr filter Signed-off-by: Paul B Mahol --- 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