[FFmpeg-devel] avfilter: add arbitrary audio IIR filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi         |  14 +++
 libavfilter/Makefile     |   1 +
 libavfilter/af_aiir.c    | 232 +++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 4 files changed, 248 insertions(+)
 create mode 100644 libavfilter/af_aiir.c

On 2 January 2018 at 16:18, Paul B Mahol <onemda@gmail.com> wrote:

> Signed-off-by: Paul B Mahol <onemda@gmail.com>
> ---
>  doc/filters.texi         |  14 +++
>  libavfilter/Makefile     |   1 +
>  libavfilter/af_aiir.c    | 232 ++++++++++++++++++++++++++++++
> +++++++++++++++++
>  libavfilter/allfilters.c |   1 +
>  4 files changed, 248 insertions(+)
>  create mode 100644 libavfilter/af_aiir.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index f651f1234d..ff911ad92e 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -1059,6 +1059,20 @@ the reduction.
>  Default is @code{average}. Can be @code{average} or @code{maximum}.
>  @end table
>
> +@section aiir
> +
> +Apply an arbitrary Infinite Impulse Response filter.
> +
> +It accepts the following parameters:
> +
> +@table @option
> +@item a
> +Set denominator coefficients.
> +
> +@item b
> +Set nominator coefficients.
> +@end table
> +
>  @section alimiter
>
>  The limiter prevents an input signal from rising over a desired threshold.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 8bde542163..1fe58ed3d2 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -43,6 +43,7 @@ OBJS-$(CONFIG_AFFTFILT_FILTER)               +=
> af_afftfilt.o
>  OBJS-$(CONFIG_AFIR_FILTER)                   += af_afir.o
>  OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
>  OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o
> +OBJS-$(CONFIG_AIIR_FILTER)                   += af_aiir.o
>  OBJS-$(CONFIG_AINTERLEAVE_FILTER)            += f_interleave.o
>  OBJS-$(CONFIG_ALIMITER_FILTER)               += af_alimiter.o
>  OBJS-$(CONFIG_ALLPASS_FILTER)                += af_biquads.o
> diff --git a/libavfilter/af_aiir.c b/libavfilter/af_aiir.c
> new file mode 100644
> index 0000000000..d1be9afa5e
> --- /dev/null
> +++ b/libavfilter/af_aiir.c
> @@ -0,0 +1,232 @@
> +/*
> + * Copyright (c) 2018 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/avassert.h"
> +#include "libavutil/avstring.h"
> +#include "libavutil/opt.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "internal.h"
> +
> +typedef struct AudioIIRContext {
> +    const AVClass *class;
> +    char *a_str, *b_str;
> +
> +    int nb_a, nb_b;
> +    double *a, *b;
> +    AVFrame *input, *output;
> +} AudioIIRContext;
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> +    AVFilterFormats *formats;
> +    AVFilterChannelLayouts *layouts;
> +    static const enum AVSampleFormat sample_fmts[] = {
> +        AV_SAMPLE_FMT_DBLP,
> +        AV_SAMPLE_FMT_NONE
> +    };
> +    int ret;
> +
> +    layouts = ff_all_channel_counts();
> +    if (!layouts)
> +        return AVERROR(ENOMEM);
> +    ret = ff_set_common_channel_layouts(ctx, layouts);
> +    if (ret < 0)
> +        return ret;
> +
> +    formats = ff_make_format_list(sample_fmts);
> +    if (!formats)
> +        return AVERROR(ENOMEM);
> +    ret = ff_set_common_formats(ctx, formats);
> +    if (ret < 0)
> +        return ret;
> +
> +    formats = ff_all_samplerates();
> +    if (!formats)
> +        return AVERROR(ENOMEM);
> +    return ff_set_common_samplerates(ctx, formats);
> +}
> +
> +static int config_output(AVFilterLink *outlink)
> +{
> +    AVFilterContext *ctx = outlink->src;
> +    AudioIIRContext *s = ctx->priv;
> +    AVFilterLink *inlink = ctx->inputs[0];
> +
> +    s->input  = ff_get_audio_buffer(inlink, s->nb_b);
> +    s->output = ff_get_audio_buffer(inlink, s->nb_a);
> +    if (!s->input || !s->output)
> +            return AVERROR(ENOMEM);
> +    return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> +    AVFilterContext  *ctx = inlink->dst;
> +    AudioIIRContext *s     = ctx->priv;
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    AVFrame *out;
> +    int ch, n;
> +
> +    if (av_frame_is_writable(in)) {
> +        out = in;
> +    } else {
> +        out = ff_get_audio_buffer(outlink, in->nb_samples);
> +        if (!out) {
> +            av_frame_free(&in);
> +            return AVERROR(ENOMEM);
> +        }
> +        av_frame_copy_props(out, in);
> +    }
> +
> +    for (ch = 0; ch < out->channels; ch++) {
> +        const double *src = (const double *)in->extended_data[ch];
> +        double *ic = (double *)s->input->extended_data[ch];
> +        double *oc = (double *)s->output->extended_data[ch];
> +        double *dst = (double *)out->extended_data[ch];
> +        const double *a = s->a;
> +        const double *b = s->b;
> +
> +        for (n = 0; n < in->nb_samples; n++) {
> +            double sample = 0.;
> +            int x;
> +
> +            memmove(&ic[1], &ic[0], (s->nb_b - 1) * sizeof(*ic));
> +            memmove(&oc[1], &oc[0], (s->nb_a - 1) * sizeof(*oc));
> +            ic[0] = src[n];
> +            for (x = 0; x < s->nb_b; x++)
> +                sample += b[x] * ic[x];
> +
> +            for (x = 1; x < s->nb_a; x++)
> +                sample -= a[x] * oc[x];
> +
> +            oc[0] = dst[n] = sample;
> +        }
> +    }
> +
> +    if (in != out)
> +        av_frame_free(&in);
> +
> +    return ff_filter_frame(outlink, out);
> +}
> +
> +static void count_items(char *item_str, int *nb_items)
> +{
> +    char *p;
> +
> +    *nb_items = 1;
> +    for (p = item_str; *p; p++) {
> +        if (*p == ' ' || *p == '|')
> +            (*nb_items)++;
> +    }
> +}
> +
> +static int read_items(char *item_str, int nb_items, double *dst)
> +{
> +    char *p, *arg, *saveptr = NULL;
> +    int i;
> +
> +    p = item_str;
> +    for (i = 0; i < nb_items; i++) {
> +        if (!(arg = av_strtok(p, " |", &saveptr)))
> +            break;
> +
> +        p = NULL;
> +        sscanf(arg, "%lf", &dst[i]);
> +    }
> +
> +    return 0;
> +}
> +
> +static av_cold int init(AVFilterContext *ctx)
> +{
> +    AudioIIRContext *s = ctx->priv;
> +    int i;
> +
> +    count_items(s->a_str, &s->nb_a);
> +    count_items(s->b_str, &s->nb_b);
> +
> +    s->a = av_calloc(s->nb_a, sizeof(*s->a));
> +    s->b = av_calloc(s->nb_b, sizeof(*s->b));
> +    if (!s->a || !s->b)
> +        return AVERROR(ENOMEM);
> +
> +    read_items(s->a_str, s->nb_a, s->a);
> +    read_items(s->b_str, s->nb_b, s->b);
> +
> +    for (i = 1; i < s->nb_a; i++)
> +        s->a[i] /= s->a[0];
> +
> +    for (i = 0; i < s->nb_b; i++)
> +        s->b[i] /= s->a[0];
> +
> +    return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> +    AudioIIRContext *s = ctx->priv;
> +
> +    av_freep(&s->a);
> +    av_freep(&s->b);
> +    av_frame_free(&s->input);
> +    av_frame_free(&s->output);
> +}
> +
> +static const AVFilterPad inputs[] = {
> +    {
> +        .name         = "default",
> +        .type         = AVMEDIA_TYPE_AUDIO,
> +        .filter_frame = filter_frame,
> +    },
> +    { NULL }
> +};
> +
> +static const AVFilterPad outputs[] = {
> +    {
> +        .name         = "default",
> +        .type         = AVMEDIA_TYPE_AUDIO,
> +        .config_props = config_output,
> +    },
> +    { NULL }
> +};
> +
> +#define OFFSET(x) offsetof(AudioIIRContext, x)
> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption aiir_options[] = {
> +    { "a", "set A coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING,
> {.str="1 1"}, 0, 0, .flags = FLAGS },
> +    { "b", "set B coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING,
> {.str="1 1"}, 0, 0, .flags = FLAGS },
> +    { NULL },
> +};
> +
> +AVFILTER_DEFINE_CLASS(aiir);
> +
> +AVFilter ff_af_aiir = {
> +    .name          = "aiir",
> +    .description   = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse
> Response filter with supplied coefficients."),
> +    .priv_size     = sizeof(AudioIIRContext),
> +    .init          = init,
> +    .uninit        = uninit,
> +    .query_formats = query_formats,
> +    .inputs        = inputs,
> +    .outputs       = outputs,
> +    .priv_class    = &aiir_class,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 67c073091f..705c03c22c 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -54,6 +54,7 @@ static void register_all(void)
>      REGISTER_FILTER(AFIR,           afir,           af);
>      REGISTER_FILTER(AFORMAT,        aformat,        af);
>      REGISTER_FILTER(AGATE,          agate,          af);
> +    REGISTER_FILTER(AIIR,           aiir,           af);
>      REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);
>      REGISTER_FILTER(ALIMITER,       alimiter,       af);
>      REGISTER_FILTER(ALLPASS,        allpass,        af);
> --
> 2.11.0
>
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>

lavc has an IIR filter (libavcodec/iirfilter.h), couldn't you reuse it?

On 1/2/18, Rostislav Pehlivanov <atomnuker@gmail.com> wrote:
>
> lavc has an IIR filter (libavcodec/iirfilter.h), couldn't you reuse it?

No, because its using floats, its more limited and I do not want to depend
on lavc for such simple filter.