avfilter: add audio signal to distortion ratio filter

pull/370/head
Paul B Mahol 3 years ago
parent 30d4609484
commit 8f26ebde14
  1. 1
      Changelog
  2. 7
      doc/filters.texi
  3. 1
      libavfilter/Makefile
  4. 172
      libavfilter/af_asdr.c
  5. 1
      libavfilter/allfilters.c
  6. 2
      libavfilter/version.h

@ -24,6 +24,7 @@ version <next>:
- amr parser
- (a)latency filters
- GEM Raster image decoder
- asdr audio filter
version 4.4:

@ -2556,6 +2556,13 @@ noise removed from input signal.
This filter supports the all above options as @ref{commands}.
@section asdr
Measure Audio 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 asetnsamples
Set the number of samples per each output audio frame.

@ -81,6 +81,7 @@ OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_AREVERSE_FILTER) += f_reverse.o
OBJS-$(CONFIG_ARNNDN_FILTER) += af_arnndn.o
OBJS-$(CONFIG_ASDR_FILTER) += af_asdr.o
OBJS-$(CONFIG_ASEGMENT_FILTER) += f_segment.o
OBJS-$(CONFIG_ASELECT_FILTER) += f_select.o
OBJS-$(CONFIG_ASENDCMD_FILTER) += f_sendcmd.o

@ -0,0 +1,172 @@
/*
* Copyright (c) 2021 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "filters.h"
#include "internal.h"
typedef struct AudioSDRContext {
int channels;
int64_t pts;
double *sum_u;
double *sum_uv;
AVFrame *cache[2];
} AudioSDRContext;
static void sdr(AVFilterContext *ctx, const AVFrame *u, const AVFrame *v)
{
AudioSDRContext *s = ctx->priv;
for (int ch = 0; ch < u->channels; ch++) {
const double *const us = (double *)u->extended_data[ch];
const double *const vs = (double *)v->extended_data[ch];
double sum_uv = s->sum_uv[ch];
double sum_u = s->sum_u[ch];
for (int n = 0; n < u->nb_samples; n++) {
sum_u += us[n] * us[n];
sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]);
}
s->sum_uv[ch] = sum_uv;
s->sum_u[ch] = sum_u;
}
}
static int activate(AVFilterContext *ctx)
{
AudioSDRContext *s = ctx->priv;
int ret, status;
int available;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
available = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), ff_inlink_queued_samples(ctx->inputs[1]));
if (available > 0) {
AVFrame *out;
for (int i = 0; i < 2; i++) {
ret = ff_inlink_consume_samples(ctx->inputs[i], available, available, &s->cache[i]);
if (ret > 0) {
if (s->pts == AV_NOPTS_VALUE)
s->pts = s->cache[i]->pts;
}
}
sdr(ctx, s->cache[0], s->cache[1]);
av_frame_free(&s->cache[1]);
out = s->cache[0];
out->nb_samples = available;
out->pts = s->pts;
s->pts += available;
s->cache[0] = NULL;
return ff_filter_frame(ctx->outputs[0], out);
}
for (int i = 0; i < 2; i++) {
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
ff_outlink_set_status(ctx->outputs[0], status, s->pts);
return 0;
}
}
if (ff_outlink_frame_wanted(ctx->outputs[0])) {
for (int i = 0; i < 2; i++) {
if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
continue;
ff_inlink_request_frame(ctx->inputs[i]);
}
return 0;
}
return FFERROR_NOT_READY;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AudioSDRContext *s = ctx->priv;
s->pts = AV_NOPTS_VALUE;
s->channels = inlink->channels;
s->sum_u = av_calloc(outlink->channels, sizeof(*s->sum_u));
s->sum_uv = av_calloc(outlink->channels, sizeof(*s->sum_uv));
if (!s->sum_u || !s->sum_uv)
return AVERROR(ENOMEM);
return 0;
}
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]));
av_frame_free(&s->cache[0]);
av_frame_free(&s->cache[1]);
av_freep(&s->sum_u);
av_freep(&s->sum_uv);
}
static const AVFilterPad inputs[] = {
{
.name = "input0",
.type = AVMEDIA_TYPE_AUDIO,
},
{
.name = "input1",
.type = AVMEDIA_TYPE_AUDIO,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_asdr = {
.name = "asdr",
.description = NULL_IF_CONFIG_SMALL("Measure Audio Signal-to-Distortion Ratio."),
.priv_size = sizeof(AudioSDRContext),
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
};

@ -74,6 +74,7 @@ extern const AVFilter ff_af_arealtime;
extern const AVFilter ff_af_aresample;
extern const AVFilter ff_af_areverse;
extern const AVFilter ff_af_arnndn;
extern const AVFilter ff_af_asdr;
extern const AVFilter ff_af_asegment;
extern const AVFilter ff_af_aselect;
extern const AVFilter ff_af_asendcmd;

@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 8
#define LIBAVFILTER_VERSION_MINOR 11
#define LIBAVFILTER_VERSION_MINOR 12
#define LIBAVFILTER_VERSION_MICRO 100

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