[FFmpeg-cvslog] avfilter: add arbitrary audio IIR filter
Paul B Mahol
git at videolan.org
Fri Jan 5 18:07:27 EET 2018
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Tue Jan 2 14:30:54 2018 +0100| [7bb1be9af0ea41d6f342655e1d15e30f662fe0f3] | committer: Paul B Mahol
avfilter: add arbitrary audio IIR filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7bb1be9af0ea41d6f342655e1d15e30f662fe0f3
---
Changelog | 1 +
doc/filters.texi | 35 +++++
libavfilter/Makefile | 1 +
libavfilter/af_aiir.c | 334 +++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/af_biquads.c | 2 +
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
7 files changed, 375 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 3d966c202b..56a3d684d6 100644
--- a/Changelog
+++ b/Changelog
@@ -33,6 +33,7 @@ version <next>:
- deconvolve video filter
- entropy video filter
- hilbert audio filter source
+- aiir audio filter
version 3.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index 73537c524f..1a2a93b97a 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1060,6 +1060,41 @@ the reduction.
Default is @code{average}. Can be @code{average} or @code{maximum}.
@end table
+ at section aiir
+
+Apply an arbitrary Infinite Impulse Response filter.
+
+It accepts the following parameters:
+
+ at table @option
+ at item a
+Set denominator/poles coefficients.
+
+ at item b
+Set nominator/zeros coefficients.
+
+ at item dry_gain
+Set input gain.
+
+ at item wet_gain
+Set output gain.
+ at end table
+
+Coefficients are separated by spaces and are in ascending order.
+Different coefficients can be provided for every channel, in such case
+use '|' to separate coefficients. Last provided coefficients will be
+used for all remaining channels.
+
+ at subsection Examples
+
+ at itemize
+ at item
+Apply 2 pole elliptic notch at arround 5000Hz for 48000 Hz sample rate:
+ at example
+aiir=b=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:a=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1
+ at end example
+ at end itemize
+
@section alimiter
The limiter prevents an input signal from rising over a desired threshold.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8a103d4f33..256dfabd66 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..29010bde29
--- /dev/null
+++ b/libavfilter/af_aiir.c
@@ -0,0 +1,334 @@
+/*
+ * 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;
+ double dry_gain, wet_gain;
+
+ int *nb_a, *nb_b;
+ double **a, **b;
+ double **input, **output;
+ int clippings;
+ int channels;
+
+ void (*iir_frame)(AVFilterContext *ctx, AVFrame *in, AVFrame *out);
+} AudioIIRContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_S16P,
+ 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);
+}
+
+#define IIR_FRAME(name, type, min, max, need_clipping) \
+static void iir_frame_## name(AVFilterContext *ctx, AVFrame *in, AVFrame *out) \
+{ \
+ AudioIIRContext *s = ctx->priv; \
+ const double ig = s->dry_gain; \
+ const double og = s->wet_gain; \
+ int ch, n; \
+ \
+ for (ch = 0; ch < out->channels; ch++) { \
+ const type *src = (const type *)in->extended_data[ch]; \
+ double *ic = (double *)s->input[ch]; \
+ double *oc = (double *)s->output[ch]; \
+ const int nb_a = s->nb_a[ch]; \
+ const int nb_b = s->nb_b[ch]; \
+ const double *a = s->a[ch]; \
+ const double *b = s->b[ch]; \
+ type *dst = (type *)out->extended_data[ch]; \
+ \
+ for (n = 0; n < in->nb_samples; n++) { \
+ double sample = 0.; \
+ int x; \
+ \
+ memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
+ memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
+ ic[0] = src[n] * ig; \
+ for (x = 0; x < nb_b; x++) \
+ sample += b[x] * ic[x]; \
+ \
+ for (x = 1; x < nb_a; x++) \
+ sample -= a[x] * oc[x]; \
+ \
+ oc[0] = sample; \
+ sample *= og; \
+ if (need_clipping && sample < min) { \
+ s->clippings++; \
+ dst[n] = min; \
+ } else if (need_clipping && sample > max) { \
+ s->clippings++; \
+ dst[n] = max; \
+ } else { \
+ dst[n] = sample; \
+ } \
+ } \
+ } \
+}
+
+IIR_FRAME(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
+IIR_FRAME(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
+IIR_FRAME(fltp, float, -1., 1., 0)
+IIR_FRAME(dblp, double, -1., 1., 0)
+
+static void count_coefficients(char *item_str, int *nb_items)
+{
+ char *p;
+
+ *nb_items = 1;
+ for (p = item_str; *p && *p != '|'; p++) {
+ if (*p == ' ')
+ (*nb_items)++;
+ }
+}
+
+static int read_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
+{
+ char *p, *arg, *old_str, *saveptr = NULL;
+ int i;
+
+ p = old_str = av_strdup(item_str);
+ if (!p)
+ return AVERROR(ENOMEM);
+ for (i = 0; i < nb_items; i++) {
+ if (!(arg = av_strtok(p, " ", &saveptr)))
+ break;
+
+ p = NULL;
+ if (sscanf(arg, "%lf", &dst[i]) != 1) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
+ return AVERROR(EINVAL);
+ }
+ }
+
+ av_freep(&old_str);
+
+ return 0;
+}
+
+static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache)
+{
+ char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
+ int i, ret;
+
+ p = old_str = av_strdup(item_str);
+ if (!p)
+ return AVERROR(ENOMEM);
+ for (i = 0; i < channels; i++) {
+ if (!(arg = av_strtok(p, "|", &saveptr)))
+ arg = prev_arg;
+
+ p = NULL;
+ count_coefficients(arg, &nb[i]);
+ cache[i] = av_calloc(nb[i], sizeof(cache[i]));
+ c[i] = av_calloc(nb[i], sizeof(c[i]));
+ if (!c[i] || !cache[i])
+ return AVERROR(ENOMEM);
+
+ ret = read_coefficients(ctx, arg, nb[i], c[i]);
+ if (ret < 0)
+ return ret;
+ prev_arg = arg;
+ }
+
+ av_freep(&old_str);
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioIIRContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ int ch, ret, i;
+
+ s->channels = inlink->channels;
+ s->a = av_calloc(inlink->channels, sizeof(*s->a));
+ s->b = av_calloc(inlink->channels, sizeof(*s->b));
+ s->nb_a = av_calloc(inlink->channels, sizeof(*s->nb_a));
+ s->nb_b = av_calloc(inlink->channels, sizeof(*s->nb_b));
+ s->input = av_calloc(inlink->channels, sizeof(*s->input));
+ s->output = av_calloc(inlink->channels, sizeof(*s->output));
+ if (!s->a || !s->b || !s->nb_a || !s->nb_b || !s->input || !s->output)
+ return AVERROR(ENOMEM);
+
+ ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output);
+ if (ret < 0)
+ return ret;
+
+ ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input);
+ if (ret < 0)
+ return ret;
+
+ for (ch = 0; ch < inlink->channels; ch++) {
+ for (i = 1; i < s->nb_a[ch]; i++) {
+ s->a[ch][i] /= s->a[ch][0];
+ }
+
+ for (i = 0; i < s->nb_b[ch]; i++) {
+ s->b[ch][i] /= s->a[ch][0];
+ }
+ }
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_DBLP: s->iir_frame = iir_frame_dblp; break;
+ case AV_SAMPLE_FMT_FLTP: s->iir_frame = iir_frame_fltp; break;
+ case AV_SAMPLE_FMT_S32P: s->iir_frame = iir_frame_s32p; break;
+ case AV_SAMPLE_FMT_S16P: s->iir_frame = iir_frame_s16p; break;
+ }
+
+ 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;
+
+ 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);
+ }
+
+ s->iir_frame(ctx, in, out);
+
+ if (s->clippings > 0)
+ av_log(ctx, AV_LOG_WARNING, "clipping %d times. Please reduce gain.\n", s->clippings);
+ s->clippings = 0;
+
+ if (in != out)
+ av_frame_free(&in);
+
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioIIRContext *s = ctx->priv;
+ int ch;
+
+ if (s->a) {
+ for (ch = 0; ch < s->channels; ch++) {
+ av_freep(&s->a[ch]);
+ av_freep(&s->output[ch]);
+ }
+ }
+ av_freep(&s->a);
+
+ if (s->b) {
+ for (ch = 0; ch < s->channels; ch++) {
+ av_freep(&s->b[ch]);
+ av_freep(&s->input[ch]);
+ }
+ }
+ av_freep(&s->b);
+
+ av_freep(&s->input);
+ av_freep(&s->output);
+
+ av_freep(&s->nb_a);
+ av_freep(&s->nb_b);
+}
+
+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 AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aiir_options[] = {
+ { "a", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
+ { "b", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
+ { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
+ { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
+ { 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),
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = inputs,
+ .outputs = outputs,
+ .priv_class = &aiir_class,
+};
diff --git a/libavfilter/af_biquads.c b/libavfilter/af_biquads.c
index b0772b9fdc..6e60e3b1b7 100644
--- a/libavfilter/af_biquads.c
+++ b/libavfilter/af_biquads.c
@@ -375,6 +375,8 @@ static int config_filter(AVFilterLink *outlink, int reset)
av_assert0(0);
}
+ av_log(ctx, AV_LOG_VERBOSE, "a=%lf %lf %lf:b=%lf %lf %lf\n", s->a0, s->a1, s->a2, s->b0, s->b1, s->b2);
+
s->a1 /= s->a0;
s->a2 /= s->a0;
s->b0 /= s->a0;
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 8c4ed6bd03..753ae968aa 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);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index ac8bec4cb8..c07f4d30d9 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 10
+#define LIBAVFILTER_VERSION_MINOR 11
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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