Paul B Mahol (12021-09-12):
> Signed-off-by: Paul B Mahol <one...@gmail.com>
> ---
> 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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