On 2 January 2018 at 16:18, Paul B Mahol <one...@gmail.com> wrote:
> Signed-off-by: Paul B Mahol <one...@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
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
lavc has an IIR filter (libavcodec/iirfilter.h), couldn't you reuse it?
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