| /* |
| * 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 <float.h> |
| |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/common.h" |
| #include "libavutil/mem.h" |
| |
| #include "avfilter.h" |
| #include "filters.h" |
| #include "internal.h" |
| |
| typedef struct ChanStats { |
| double u; |
| double v; |
| double uv; |
| } ChanStats; |
| |
| typedef struct AudioSDRContext { |
| int channels; |
| uint64_t nb_samples; |
| double max; |
| |
| ChanStats *chs; |
| |
| AVFrame *cache[2]; |
| |
| int (*filter)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); |
| } AudioSDRContext; |
| |
| #define SDR_FILTER(name, type) \ |
| static int sdr_##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++) { \ |
| ChanStats *chs = &s->chs[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.; \ |
| \ |
| for (int n = 0; n < nb_samples; n++) { \ |
| sum_u += us[n] * us[n]; \ |
| sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \ |
| } \ |
| \ |
| chs->uv += sum_uv; \ |
| chs->u += sum_u; \ |
| } \ |
| \ |
| return 0; \ |
| } |
| |
| 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++) { \ |
| ChanStats *chs = &s->chs[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]; \ |
| } \ |
| \ |
| chs->uv += sum_uv; \ |
| chs->u += sum_u; \ |
| chs->v += 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)\ |
| { \ |
| 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++) { \ |
| ChanStats *chs = &s->chs[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]); \ |
| \ |
| chs->uv += sum_uv; \ |
| } \ |
| \ |
| return 0; \ |
| } |
| |
| PSNR_FILTER(fltp, float) |
| PSNR_FILTER(dblp, double) |
| |
| static int activate(AVFilterContext *ctx) |
| { |
| AudioSDRContext *s = ctx->priv; |
| AVFilterLink *outlink = ctx->outputs[0]; |
| int ret, status, available; |
| int64_t pts; |
| |
| FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, 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) { |
| av_frame_free(&s->cache[0]); |
| av_frame_free(&s->cache[1]); |
| return ret; |
| } |
| } |
| |
| if (!ctx->is_disabled) |
| ff_filter_execute(ctx, s->filter, NULL, NULL, |
| FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); |
| |
| av_frame_free(&s->cache[1]); |
| out = s->cache[0]; |
| s->cache[0] = NULL; |
| |
| s->nb_samples += available; |
| return ff_filter_frame(outlink, out); |
| } |
| |
| for (int i = 0; i < 2; i++) { |
| if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { |
| ff_outlink_set_status(outlink, status, pts); |
| return 0; |
| } |
| } |
| |
| if (ff_outlink_frame_wanted(outlink)) { |
| for (int i = 0; i < 2; i++) { |
| if (s->cache[i] || 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->channels = inlink->ch_layout.nb_channels; |
| |
| 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->chs = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->chs)); |
| if (!s->chs) |
| return AVERROR(ENOMEM); |
| |
| return 0; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| AudioSDRContext *s = ctx->priv; |
| |
| 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, 10. * log10(s->chs[ch].u / s->chs[ch].uv)); |
| } else if (!strcmp(ctx->filter->name, "asisdr")) { |
| for (int ch = 0; ch < s->channels; ch++) { |
| double scale = s->chs[ch].uv / s->chs[ch].v; |
| 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); |
| |
| 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->chs[ch].uv > 0.0 ? 2.0 * log(s->max) - log(s->nb_samples / s->chs[ch].uv) : 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]); |
| |
| av_freep(&s->chs); |
| } |
| |
| 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, |
| .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), |
| }; |
| |
| 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), |
| }; |
| |
| 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), |
| }; |