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
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