Paul B Mahol (12021-09-12): > Signed-off-by: Paul B Mahol <[email protected]> > --- > doc/filters.texi | 7 ++ > libavfilter/Makefile | 1 + > libavfilter/af_asdr.c | 197 +++++++++++++++++++++++++++++++++++++++ > libavfilter/allfilters.c | 1 + > 4 files changed, 206 insertions(+) > create mode 100644 libavfilter/af_asdr.c > > diff --git a/doc/filters.texi b/doc/filters.texi > index 8f20ccf8c6..6af7344820 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -2531,6 +2531,13 @@ 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. > diff --git a/libavfilter/Makefile b/libavfilter/Makefile > index 76c65c3f42..865252ef3f 100644 > --- a/libavfilter/Makefile > +++ b/libavfilter/Makefile > @@ -82,6 +82,7 @@ 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 > diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c > new file mode 100644 > index 0000000000..25032445cd > --- /dev/null > +++ b/libavfilter/af_asdr.c > @@ -0,0 +1,197 @@ > +/* > + * 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 int query_formats(AVFilterContext *ctx) > +{ > + static const enum AVSampleFormat sample_fmts[] = { > + AV_SAMPLE_FMT_DBLP, > + AV_SAMPLE_FMT_NONE > + }; > + int ret = ff_set_common_all_channel_counts(ctx); > + if (ret < 0) > + return ret; > + > + ret = ff_set_common_formats_from_list(ctx, sample_fmts); > + if (ret < 0) > + return ret; > + > + return ff_set_common_all_samplerates(ctx); > +} > + > +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); > + }
Here, you need an else for the case where one input has samples, to call
ff_inlink_request_frame().
> +
> + 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, pts);
> + return 0;
> + }
> + }
> +
> + if (ff_inlink_queued_samples(ctx->inputs[0]) > 0 &&
> + ff_inlink_queued_samples(ctx->inputs[1]) > 0) {
This condition can never be true, since you just consumed all the
samples from one of the inputs.
> + ff_filter_set_ready(ctx, 10);
> + 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;
> + outlink->format = inlink->format;
> + outlink->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),
> + .query_formats = query_formats,
> + .activate = activate,
> + .uninit = uninit,
> + FILTER_INPUTS(inputs),
> + FILTER_OUTPUTS(outputs),
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 73a0bf9c44..7234ca6dbe 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -75,6 +75,7 @@ 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;
Regards,
--
Nicolas George
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