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avfilter: add arbitrary audio IIR filter 

Signed-off-by: Paul B Mahol <onemda@gmail.com>
tags/n4.0
Paul B Mahol 7 years ago
parent
b2be76c0a4
commit
7bb1be9af0
7 changed files with 375 additions and 1 deletions
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  1. +1
    -0
      Changelog
  2. +35
    -0
      doc/filters.texi
  3. +1
    -0
      libavfilter/Makefile
  4. +334
    -0
      libavfilter/af_aiir.c
  5. +2
    -0
      libavfilter/af_biquads.c
  6. +1
    -0
      libavfilter/allfilters.c
  7. +1
    -1
      libavfilter/version.h

+ 1
- 0
Changelog View File

@@ -33,6 +33,7 @@ version <next>:
- deconvolve video filter
- entropy video filter
- hilbert audio filter source
- aiir audio filter


version 3.4:


+ 35
- 0
doc/filters.texi View File

@@ -1060,6 +1060,41 @@ 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/poles coefficients.

@item b
Set nominator/zeros coefficients.

@item dry_gain
Set input gain.

@item wet_gain
Set output gain.
@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.

@subsection Examples

@itemize
@item
Apply 2 pole elliptic notch at arround 5000Hz for 48000 Hz sample rate:
@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
@end example
@end itemize

@section alimiter

The limiter prevents an input signal from rising over a desired threshold.


+ 1
- 0
libavfilter/Makefile View File

@@ -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


+ 334
- 0
libavfilter/af_aiir.c View File

@@ -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,
};

+ 2
- 0
libavfilter/af_biquads.c View File

@@ -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;


+ 1
- 0
libavfilter/allfilters.c View File

@@ -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);


+ 1
- 1
libavfilter/version.h View File

@@ -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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