[FFmpeg-cvslog] avfilter: add apsnr filter
Paul B Mahol
git at videolan.org
Mon Aug 14 12:35:06 EEST 2023
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun Aug 13 02:57:57 2023 +0200| [951def850abe9dc77311e5afd4b581defa1575bb] | committer: Paul B Mahol
avfilter: add apsnr filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=951def850abe9dc77311e5afd4b581defa1575bb
---
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 43e9c037b9..5d1f10f95d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2836,6 +2836,13 @@ Default value is 8.
This filter supports the all above options as @ref{commands}.
+ at 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 30a0e22ef8..9cd1407250 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -84,6 +84,7 @@ 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 089ad3a0ed..f6017c41c9 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -70,6 +70,7 @@ 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;
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