mirror of https://git.videolan.org/git/ffmpeg.git
314 lines
13 KiB
C
314 lines
13 KiB
C
/*
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* Copyright (c) 2021 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <float.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/mem.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "internal.h"
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typedef struct ChanStats {
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double u;
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double v;
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double uv;
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} ChanStats;
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typedef struct AudioSDRContext {
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int channels;
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uint64_t nb_samples;
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double max;
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ChanStats *chs;
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AVFrame *cache[2];
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int (*filter)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
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} AudioSDRContext;
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#define SDR_FILTER(name, type) \
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static int sdr_##name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)\
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{ \
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AudioSDRContext *s = ctx->priv; \
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AVFrame *u = s->cache[0]; \
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AVFrame *v = s->cache[1]; \
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const int channels = u->ch_layout.nb_channels; \
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const int start = (channels * jobnr) / nb_jobs; \
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const int end = (channels * (jobnr+1)) / nb_jobs; \
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const int nb_samples = u->nb_samples; \
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\
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for (int ch = start; ch < end; ch++) { \
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ChanStats *chs = &s->chs[ch]; \
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const type *const us = (type *)u->extended_data[ch]; \
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const type *const vs = (type *)v->extended_data[ch]; \
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double sum_uv = 0.; \
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double sum_u = 0.; \
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\
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for (int n = 0; n < nb_samples; n++) { \
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sum_u += us[n] * us[n]; \
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sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
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} \
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\
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chs->uv += sum_uv; \
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chs->u += sum_u; \
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} \
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\
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return 0; \
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}
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SDR_FILTER(fltp, float)
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SDR_FILTER(dblp, double)
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#define SISDR_FILTER(name, type) \
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static int sisdr_##name(AVFilterContext *ctx, void *arg,int jobnr,int nb_jobs)\
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{ \
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AudioSDRContext *s = ctx->priv; \
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AVFrame *u = s->cache[0]; \
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AVFrame *v = s->cache[1]; \
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const int channels = u->ch_layout.nb_channels; \
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const int start = (channels * jobnr) / nb_jobs; \
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const int end = (channels * (jobnr+1)) / nb_jobs; \
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const int nb_samples = u->nb_samples; \
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\
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for (int ch = start; ch < end; ch++) { \
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ChanStats *chs = &s->chs[ch]; \
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const type *const us = (type *)u->extended_data[ch]; \
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const type *const vs = (type *)v->extended_data[ch]; \
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double sum_uv = 0.; \
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double sum_u = 0.; \
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double sum_v = 0.; \
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\
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for (int n = 0; n < nb_samples; n++) { \
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sum_u += us[n] * us[n]; \
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sum_v += vs[n] * vs[n]; \
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sum_uv += us[n] * vs[n]; \
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} \
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\
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chs->uv += sum_uv; \
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chs->u += sum_u; \
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chs->v += sum_v; \
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} \
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\
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return 0; \
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}
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SISDR_FILTER(fltp, float)
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SISDR_FILTER(dblp, double)
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#define PSNR_FILTER(name, type) \
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static int psnr_##name(AVFilterContext *ctx, void *arg, int jobnr,int nb_jobs)\
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{ \
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AudioSDRContext *s = ctx->priv; \
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AVFrame *u = s->cache[0]; \
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AVFrame *v = s->cache[1]; \
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const int channels = u->ch_layout.nb_channels; \
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const int start = (channels * jobnr) / nb_jobs; \
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const int end = (channels * (jobnr+1)) / nb_jobs; \
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const int nb_samples = u->nb_samples; \
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\
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for (int ch = start; ch < end; ch++) { \
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ChanStats *chs = &s->chs[ch]; \
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const type *const us = (type *)u->extended_data[ch]; \
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const type *const vs = (type *)v->extended_data[ch]; \
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double sum_uv = 0.; \
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\
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for (int n = 0; n < nb_samples; n++) \
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sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
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\
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chs->uv += sum_uv; \
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} \
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\
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return 0; \
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}
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PSNR_FILTER(fltp, float)
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PSNR_FILTER(dblp, double)
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static int activate(AVFilterContext *ctx)
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{
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AudioSDRContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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int ret, status, available;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
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available = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), ff_inlink_queued_samples(ctx->inputs[1]));
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if (available > 0) {
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AVFrame *out;
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for (int i = 0; i < 2; i++) {
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ret = ff_inlink_consume_samples(ctx->inputs[i], available, available, &s->cache[i]);
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if (ret < 0) {
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av_frame_free(&s->cache[0]);
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av_frame_free(&s->cache[1]);
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return ret;
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}
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}
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if (!ctx->is_disabled)
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ff_filter_execute(ctx, s->filter, NULL, NULL,
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FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
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av_frame_free(&s->cache[1]);
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out = s->cache[0];
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s->cache[0] = NULL;
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s->nb_samples += available;
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return ff_filter_frame(outlink, out);
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}
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for (int i = 0; i < 2; i++) {
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
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ff_outlink_set_status(outlink, status, pts);
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return 0;
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}
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}
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if (ff_outlink_frame_wanted(outlink)) {
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for (int i = 0; i < 2; i++) {
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if (s->cache[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0)
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continue;
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ff_inlink_request_frame(ctx->inputs[i]);
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return 0;
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}
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}
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return FFERROR_NOT_READY;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AVFilterLink *inlink = ctx->inputs[0];
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AudioSDRContext *s = ctx->priv;
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s->channels = inlink->ch_layout.nb_channels;
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if (!strcmp(ctx->filter->name, "asdr"))
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s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
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else if (!strcmp(ctx->filter->name, "asisdr"))
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s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sisdr_fltp : sisdr_dblp;
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else
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s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? psnr_fltp : psnr_dblp;
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s->max = inlink->format == AV_SAMPLE_FMT_FLTP ? FLT_MAX : DBL_MAX;
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s->chs = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->chs));
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if (!s->chs)
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return AVERROR(ENOMEM);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioSDRContext *s = ctx->priv;
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if (!strcmp(ctx->filter->name, "asdr")) {
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for (int ch = 0; ch < s->channels; ch++)
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av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 10. * log10(s->chs[ch].u / s->chs[ch].uv));
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} else if (!strcmp(ctx->filter->name, "asisdr")) {
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for (int ch = 0; ch < s->channels; ch++) {
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double scale = s->chs[ch].uv / s->chs[ch].v;
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double sisdr = scale * scale * s->chs[ch].v / fmax(0., s->chs[ch].u + scale*scale*s->chs[ch].v - 2.0*scale*s->chs[ch].uv);
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av_log(ctx, AV_LOG_INFO, "SI-SDR ch%d: %g dB\n", ch, 10. * log10(sisdr));
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}
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} else {
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for (int ch = 0; ch < s->channels; ch++) {
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double psnr = s->chs[ch].uv > 0.0 ? 2.0 * log(s->max) - log(s->nb_samples / s->chs[ch].uv) : INFINITY;
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av_log(ctx, AV_LOG_INFO, "PSNR ch%d: %g dB\n", ch, psnr);
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}
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}
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av_frame_free(&s->cache[0]);
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av_frame_free(&s->cache[1]);
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av_freep(&s->chs);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "input0",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{
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.name = "input1",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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};
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const AVFilter ff_af_asdr = {
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.name = "asdr",
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.description = NULL_IF_CONFIG_SMALL("Measure Audio Signal-to-Distortion Ratio."),
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.priv_size = sizeof(AudioSDRContext),
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.activate = activate,
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.uninit = uninit,
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.flags = AVFILTER_FLAG_METADATA_ONLY |
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AVFILTER_FLAG_SLICE_THREADS |
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AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBLP),
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};
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const AVFilter ff_af_apsnr = {
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.name = "apsnr",
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.description = NULL_IF_CONFIG_SMALL("Measure Audio Peak Signal-to-Noise Ratio."),
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.priv_size = sizeof(AudioSDRContext),
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.activate = activate,
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.uninit = uninit,
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.flags = AVFILTER_FLAG_METADATA_ONLY |
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AVFILTER_FLAG_SLICE_THREADS |
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AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBLP),
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};
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const AVFilter ff_af_asisdr = {
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.name = "asisdr",
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.description = NULL_IF_CONFIG_SMALL("Measure Audio Scale-Invariant Signal-to-Distortion Ratio."),
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.priv_size = sizeof(AudioSDRContext),
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.activate = activate,
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.uninit = uninit,
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.flags = AVFILTER_FLAG_METADATA_ONLY |
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AVFILTER_FLAG_SLICE_THREADS |
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AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBLP),
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};
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