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FFmpeg/libavfilter/af_aiir.c
Paul B Mahol 1329db8cfb avfilter/af_aiir: simplify polynomial evaluation
2020-05-30 18:04:14 +02:00

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/*
* 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 <float.h>
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/opt.h"
#include "libavutil/xga_font_data.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
typedef struct Pair {
int a, b;
} Pair;
typedef struct BiquadContext {
double a[3];
double b[3];
double i1, i2;
double o1, o2;
} BiquadContext;
typedef struct IIRChannel {
int nb_ab[2];
double *ab[2];
double g;
double *cache[2];
BiquadContext *biquads;
int clippings;
} IIRChannel;
typedef struct AudioIIRContext {
const AVClass *class;
char *a_str, *b_str, *g_str;
double dry_gain, wet_gain;
double mix;
int normalize;
int format;
int process;
int precision;
int response;
int w, h;
int ir_channel;
AVRational rate;
AVFrame *video;
IIRChannel *iir;
int channels;
enum AVSampleFormat sample_format;
int (*iir_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs);
} AudioIIRContext;
static int query_formats(AVFilterContext *ctx)
{
AudioIIRContext *s = ctx->priv;
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
static const enum AVPixelFormat pix_fmts[] = {
AV_PIX_FMT_RGB0,
AV_PIX_FMT_NONE
};
int ret;
if (s->response) {
AVFilterLink *videolink = ctx->outputs[1];
formats = ff_make_format_list(pix_fmts);
if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
return ret;
}
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
sample_fmts[0] = s->sample_format;
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_CH(name, type, min, max, need_clipping) \
static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \
{ \
AudioIIRContext *s = ctx->priv; \
const double ig = s->dry_gain; \
const double og = s->wet_gain; \
const double mix = s->mix; \
ThreadData *td = arg; \
AVFrame *in = td->in, *out = td->out; \
const type *src = (const type *)in->extended_data[ch]; \
double *oc = (double *)s->iir[ch].cache[0]; \
double *ic = (double *)s->iir[ch].cache[1]; \
const int nb_a = s->iir[ch].nb_ab[0]; \
const int nb_b = s->iir[ch].nb_ab[1]; \
const double *a = s->iir[ch].ab[0]; \
const double *b = s->iir[ch].ab[1]; \
const double g = s->iir[ch].g; \
int *clippings = &s->iir[ch].clippings; \
type *dst = (type *)out->extended_data[ch]; \
int n; \
\
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 * g; \
sample = sample * mix + ic[0] * (1. - mix); \
if (need_clipping && sample < min) { \
(*clippings)++; \
dst[n] = min; \
} else if (need_clipping && sample > max) { \
(*clippings)++; \
dst[n] = max; \
} else { \
dst[n] = sample; \
} \
} \
\
return 0; \
}
IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
IIR_CH(fltp, float, -1., 1., 0)
IIR_CH(dblp, double, -1., 1., 0)
#define SERIAL_IIR_CH(name, type, min, max, need_clipping) \
static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \
{ \
AudioIIRContext *s = ctx->priv; \
const double ig = s->dry_gain; \
const double og = s->wet_gain; \
const double mix = s->mix; \
ThreadData *td = arg; \
AVFrame *in = td->in, *out = td->out; \
const type *src = (const type *)in->extended_data[ch]; \
type *dst = (type *)out->extended_data[ch]; \
IIRChannel *iir = &s->iir[ch]; \
const double g = iir->g; \
int *clippings = &iir->clippings; \
int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
int n, i; \
\
for (i = 0; i < nb_biquads; i++) { \
const double a1 = -iir->biquads[i].a[1]; \
const double a2 = -iir->biquads[i].a[2]; \
const double b0 = iir->biquads[i].b[0]; \
const double b1 = iir->biquads[i].b[1]; \
const double b2 = iir->biquads[i].b[2]; \
double i1 = iir->biquads[i].i1; \
double i2 = iir->biquads[i].i2; \
double o1 = iir->biquads[i].o1; \
double o2 = iir->biquads[i].o2; \
\
for (n = 0; n < in->nb_samples; n++) { \
double sample = ig * (i ? dst[n] : src[n]); \
double o0 = sample * b0 + i1 * b1 + i2 * b2 + o1 * a1 + o2 * a2; \
\
i2 = i1; \
i1 = src[n]; \
o2 = o1; \
o1 = o0; \
o0 *= og * g; \
\
o0 = o0 * mix + (1. - mix) * sample; \
if (need_clipping && o0 < min) { \
(*clippings)++; \
dst[n] = min; \
} else if (need_clipping && o0 > max) { \
(*clippings)++; \
dst[n] = max; \
} else { \
dst[n] = o0; \
} \
} \
iir->biquads[i].i1 = i1; \
iir->biquads[i].i2 = i2; \
iir->biquads[i].o1 = o1; \
iir->biquads[i].o2 = o2; \
} \
\
return 0; \
}
SERIAL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
SERIAL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
SERIAL_IIR_CH(fltp, float, -1., 1., 0)
SERIAL_IIR_CH(dblp, double, -1., 1., 0)
static void count_coefficients(char *item_str, int *nb_items)
{
char *p;
if (!item_str)
return;
*nb_items = 1;
for (p = item_str; *p && *p != '|'; p++) {
if (*p == ' ')
(*nb_items)++;
}
}
static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items)
{
AudioIIRContext *s = ctx->priv;
char *p, *arg, *old_str, *prev_arg = NULL, *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)))
arg = prev_arg;
if (!arg) {
av_freep(&old_str);
return AVERROR(EINVAL);
}
p = NULL;
if (sscanf(arg, "%lf", &s->iir[i].g) != 1) {
av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg);
av_freep(&old_str);
return AVERROR(EINVAL);
}
prev_arg = arg;
}
av_freep(&old_str);
return 0;
}
static int read_tf_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);
av_freep(&old_str);
return AVERROR(EINVAL);
}
}
av_freep(&old_str);
return 0;
}
static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format)
{
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, format, &dst[i*2], &dst[i*2+1]) != 2) {
av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
av_freep(&old_str);
return AVERROR(EINVAL);
}
}
av_freep(&old_str);
return 0;
}
static const char *format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd", "%lf %lfi" };
static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab)
{
AudioIIRContext *s = ctx->priv;
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++) {
IIRChannel *iir = &s->iir[i];
if (!(arg = av_strtok(p, "|", &saveptr)))
arg = prev_arg;
if (!arg) {
av_freep(&old_str);
return AVERROR(EINVAL);
}
count_coefficients(arg, &iir->nb_ab[ab]);
p = NULL;
iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double));
iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double));
if (!iir->ab[ab] || !iir->cache[ab]) {
av_freep(&old_str);
return AVERROR(ENOMEM);
}
if (s->format) {
ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]);
} else {
ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
}
if (ret < 0) {
av_freep(&old_str);
return ret;
}
prev_arg = arg;
}
av_freep(&old_str);
return 0;
}
static void cmul(double re, double im, double re2, double im2, double *RE, double *IM)
{
*RE = re * re2 - im * im2;
*IM = re * im2 + re2 * im;
}
static int expand(AVFilterContext *ctx, double *pz, int n, double *coefs)
{
coefs[2 * n] = 1.0;
for (int i = 1; i <= n; i++) {
for (int j = n - i; j < n; j++) {
double re, im;
cmul(coefs[2 * (j + 1)], coefs[2 * (j + 1) + 1],
pz[2 * (i - 1)], pz[2 * (i - 1) + 1], &re, &im);
coefs[2 * j] -= re;
coefs[2 * j + 1] -= im;
}
}
for (int i = 0; i < n + 1; i++) {
if (fabs(coefs[2 * i + 1]) > FLT_EPSILON) {
av_log(ctx, AV_LOG_ERROR, "coefs: %f of z^%d is not real; poles/zeros are not complex conjugates.\n",
coefs[2 * i + 1], i);
return AVERROR(EINVAL);
}
}
return 0;
}
static void normalize_coeffs(AVFilterContext *ctx, int ch)
{
AudioIIRContext *s = ctx->priv;
IIRChannel *iir = &s->iir[ch];
double sum_den = 0.;
if (!s->normalize)
return;
for (int i = 0; i < iir->nb_ab[1]; i++) {
sum_den += iir->ab[1][i];
}
if (sum_den > 1e-6) {
double factor, sum_num = 0.;
for (int i = 0; i < iir->nb_ab[0]; i++) {
sum_num += iir->ab[0][i];
}
factor = sum_num / sum_den;
for (int i = 0; i < iir->nb_ab[1]; i++) {
iir->ab[1][i] *= factor;
}
}
}
static int convert_zp2tf(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch, i, j, ret = 0;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
double *topc, *botc;
topc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*topc));
botc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*botc));
if (!topc || !botc) {
ret = AVERROR(ENOMEM);
goto fail;
}
ret = expand(ctx, iir->ab[0], iir->nb_ab[0], botc);
if (ret < 0) {
goto fail;
}
ret = expand(ctx, iir->ab[1], iir->nb_ab[1], topc);
if (ret < 0) {
goto fail;
}
for (j = 0, i = iir->nb_ab[1]; i >= 0; j++, i--) {
iir->ab[1][j] = topc[2 * i];
}
iir->nb_ab[1]++;
for (j = 0, i = iir->nb_ab[0]; i >= 0; j++, i--) {
iir->ab[0][j] = botc[2 * i];
}
iir->nb_ab[0]++;
normalize_coeffs(ctx, ch);
fail:
av_free(topc);
av_free(botc);
if (ret < 0)
break;
}
return ret;
}
static int decompose_zp2biquads(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch, ret;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
int current_biquad = 0;
iir->biquads = av_calloc(nb_biquads, sizeof(BiquadContext));
if (!iir->biquads)
return AVERROR(ENOMEM);
while (nb_biquads--) {
Pair outmost_pole = { -1, -1 };
Pair nearest_zero = { -1, -1 };
double zeros[4] = { 0 };
double poles[4] = { 0 };
double b[6] = { 0 };
double a[6] = { 0 };
double min_distance = DBL_MAX;
double max_mag = 0;
double factor;
int i;
for (i = 0; i < iir->nb_ab[0]; i++) {
double mag;
if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
continue;
mag = hypot(iir->ab[0][2 * i], iir->ab[0][2 * i + 1]);
if (mag > max_mag) {
max_mag = mag;
outmost_pole.a = i;
}
}
for (i = 0; i < iir->nb_ab[0]; i++) {
if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
continue;
if (iir->ab[0][2 * i ] == iir->ab[0][2 * outmost_pole.a ] &&
iir->ab[0][2 * i + 1] == -iir->ab[0][2 * outmost_pole.a + 1]) {
outmost_pole.b = i;
break;
}
}
av_log(ctx, AV_LOG_VERBOSE, "outmost_pole is %d.%d\n", outmost_pole.a, outmost_pole.b);
if (outmost_pole.a < 0 || outmost_pole.b < 0)
return AVERROR(EINVAL);
for (i = 0; i < iir->nb_ab[1]; i++) {
double distance;
if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
continue;
distance = hypot(iir->ab[0][2 * outmost_pole.a ] - iir->ab[1][2 * i ],
iir->ab[0][2 * outmost_pole.a + 1] - iir->ab[1][2 * i + 1]);
if (distance < min_distance) {
min_distance = distance;
nearest_zero.a = i;
}
}
for (i = 0; i < iir->nb_ab[1]; i++) {
if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
continue;
if (iir->ab[1][2 * i ] == iir->ab[1][2 * nearest_zero.a ] &&
iir->ab[1][2 * i + 1] == -iir->ab[1][2 * nearest_zero.a + 1]) {
nearest_zero.b = i;
break;
}
}
av_log(ctx, AV_LOG_VERBOSE, "nearest_zero is %d.%d\n", nearest_zero.a, nearest_zero.b);
if (nearest_zero.a < 0 || nearest_zero.b < 0)
return AVERROR(EINVAL);
poles[0] = iir->ab[0][2 * outmost_pole.a ];
poles[1] = iir->ab[0][2 * outmost_pole.a + 1];
zeros[0] = iir->ab[1][2 * nearest_zero.a ];
zeros[1] = iir->ab[1][2 * nearest_zero.a + 1];
if (nearest_zero.a == nearest_zero.b && outmost_pole.a == outmost_pole.b) {
zeros[2] = 0;
zeros[3] = 0;
poles[2] = 0;
poles[3] = 0;
} else {
poles[2] = iir->ab[0][2 * outmost_pole.b ];
poles[3] = iir->ab[0][2 * outmost_pole.b + 1];
zeros[2] = iir->ab[1][2 * nearest_zero.b ];
zeros[3] = iir->ab[1][2 * nearest_zero.b + 1];
}
ret = expand(ctx, zeros, 2, b);
if (ret < 0)
return ret;
ret = expand(ctx, poles, 2, a);
if (ret < 0)
return ret;
iir->ab[0][2 * outmost_pole.a] = iir->ab[0][2 * outmost_pole.a + 1] = NAN;
iir->ab[0][2 * outmost_pole.b] = iir->ab[0][2 * outmost_pole.b + 1] = NAN;
iir->ab[1][2 * nearest_zero.a] = iir->ab[1][2 * nearest_zero.a + 1] = NAN;
iir->ab[1][2 * nearest_zero.b] = iir->ab[1][2 * nearest_zero.b + 1] = NAN;
iir->biquads[current_biquad].a[0] = 1.;
iir->biquads[current_biquad].a[1] = a[2] / a[4];
iir->biquads[current_biquad].a[2] = a[0] / a[4];
iir->biquads[current_biquad].b[0] = b[4] / a[4];
iir->biquads[current_biquad].b[1] = b[2] / a[4];
iir->biquads[current_biquad].b[2] = b[0] / a[4];
if (s->normalize &&
fabs(iir->biquads[current_biquad].b[0] +
iir->biquads[current_biquad].b[1] +
iir->biquads[current_biquad].b[2]) > 1e-6) {
factor = (iir->biquads[current_biquad].a[0] +
iir->biquads[current_biquad].a[1] +
iir->biquads[current_biquad].a[2]) /
(iir->biquads[current_biquad].b[0] +
iir->biquads[current_biquad].b[1] +
iir->biquads[current_biquad].b[2]);
av_log(ctx, AV_LOG_VERBOSE, "factor=%f\n", factor);
iir->biquads[current_biquad].b[0] *= factor;
iir->biquads[current_biquad].b[1] *= factor;
iir->biquads[current_biquad].b[2] *= factor;
}
iir->biquads[current_biquad].b[0] *= (current_biquad ? 1.0 : iir->g);
iir->biquads[current_biquad].b[1] *= (current_biquad ? 1.0 : iir->g);
iir->biquads[current_biquad].b[2] *= (current_biquad ? 1.0 : iir->g);
av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n",
iir->biquads[current_biquad].a[0],
iir->biquads[current_biquad].a[1],
iir->biquads[current_biquad].a[2],
iir->biquads[current_biquad].b[0],
iir->biquads[current_biquad].b[1],
iir->biquads[current_biquad].b[2]);
current_biquad++;
}
}
return 0;
}
static void convert_pr2zp(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
int n;
for (n = 0; n < iir->nb_ab[0]; n++) {
double r = iir->ab[0][2*n];
double angle = iir->ab[0][2*n+1];
iir->ab[0][2*n] = r * cos(angle);
iir->ab[0][2*n+1] = r * sin(angle);
}
for (n = 0; n < iir->nb_ab[1]; n++) {
double r = iir->ab[1][2*n];
double angle = iir->ab[1][2*n+1];
iir->ab[1][2*n] = r * cos(angle);
iir->ab[1][2*n+1] = r * sin(angle);
}
}
}
static void convert_sp2zp(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
int n;
for (n = 0; n < iir->nb_ab[0]; n++) {
double sr = iir->ab[0][2*n];
double si = iir->ab[0][2*n+1];
double snr = 1. + sr;
double sdr = 1. - sr;
double div = sdr * sdr + si * si;
iir->ab[0][2*n] = (snr * sdr - si * si) / div;
iir->ab[0][2*n+1] = (sdr * si + snr * si) / div;
}
for (n = 0; n < iir->nb_ab[1]; n++) {
double sr = iir->ab[1][2*n];
double si = iir->ab[1][2*n+1];
double snr = 1. + sr;
double sdr = 1. - sr;
double div = sdr * sdr + si * si;
iir->ab[1][2*n] = (snr * sdr - si * si) / div;
iir->ab[1][2*n+1] = (sdr * si + snr * si) / div;
}
}
}
static void convert_pd2zp(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
int n;
for (n = 0; n < iir->nb_ab[0]; n++) {
double r = iir->ab[0][2*n];
double angle = M_PI*iir->ab[0][2*n+1]/180.;
iir->ab[0][2*n] = r * cos(angle);
iir->ab[0][2*n+1] = r * sin(angle);
}
for (n = 0; n < iir->nb_ab[1]; n++) {
double r = iir->ab[1][2*n];
double angle = M_PI*iir->ab[1][2*n+1]/180.;
iir->ab[1][2*n] = r * cos(angle);
iir->ab[1][2*n+1] = r * sin(angle);
}
}
}
static void check_stability(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
for (int n = 0; n < iir->nb_ab[0]; n++) {
double pr = hypot(iir->ab[0][2*n], iir->ab[0][2*n+1]);
if (pr >= 1.) {
av_log(ctx, AV_LOG_WARNING, "pole %d at channel %d is unstable\n", n, ch);
break;
}
}
}
}
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
{
const uint8_t *font;
int font_height;
int i;
font = avpriv_cga_font, font_height = 8;
for (i = 0; txt[i]; i++) {
int char_y, mask;
uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
for (char_y = 0; char_y < font_height; char_y++) {
for (mask = 0x80; mask; mask >>= 1) {
if (font[txt[i] * font_height + char_y] & mask)
AV_WL32(p, color);
p += 4;
}
p += pic->linesize[0] - 8 * 4;
}
}
}
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
{
int dx = FFABS(x1-x0);
int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
int err = (dx>dy ? dx : -dy) / 2, e2;
for (;;) {
AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
if (x0 == x1 && y0 == y1)
break;
e2 = err;
if (e2 >-dx) {
err -= dy;
x0--;
}
if (e2 < dy) {
err += dx;
y0 += sy;
}
}
}
static double distance(double x0, double x1, double y0, double y1)
{
return hypot(x0 - x1, y0 - y1);
}
static void get_response(int channel, int format, double w,
const double *b, const double *a,
int nb_b, int nb_a, double *magnitude, double *phase)
{
double realz, realp;
double imagz, imagp;
double real, imag;
double div;
if (format == 0) {
realz = 0., realp = 0.;
imagz = 0., imagp = 0.;
for (int x = 0; x < nb_a; x++) {
realz += cos(-x * w) * a[x];
imagz += sin(-x * w) * a[x];
}
for (int x = 0; x < nb_b; x++) {
realp += cos(-x * w) * b[x];
imagp += sin(-x * w) * b[x];
}
div = realp * realp + imagp * imagp;
real = (realz * realp + imagz * imagp) / div;
imag = (imagz * realp - imagp * realz) / div;
*magnitude = hypot(real, imag);
*phase = atan2(imag, real);
} else {
double p = 1., z = 1.;
double acc = 0.;
for (int x = 0; x < nb_a; x++) {
z *= distance(cos(w), a[2 * x], sin(w), a[2 * x + 1]);
acc += atan2(sin(w) - a[2 * x + 1], cos(w) - a[2 * x]);
}
for (int x = 0; x < nb_b; x++) {
p *= distance(cos(w), b[2 * x], sin(w), b[2 * x + 1]);
acc -= atan2(sin(w) - b[2 * x + 1], cos(w) - b[2 * x]);
}
*magnitude = z / p;
*phase = acc;
}
}
static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate)
{
AudioIIRContext *s = ctx->priv;
double *mag, *phase, *temp, *delay, min = DBL_MAX, max = -DBL_MAX;
double min_delay = DBL_MAX, max_delay = -DBL_MAX, min_phase, max_phase;
int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
char text[32];
int ch, i;
memset(out->data[0], 0, s->h * out->linesize[0]);
phase = av_malloc_array(s->w, sizeof(*phase));
temp = av_malloc_array(s->w, sizeof(*temp));
mag = av_malloc_array(s->w, sizeof(*mag));
delay = av_malloc_array(s->w, sizeof(*delay));
if (!mag || !phase || !delay || !temp)
goto end;
ch = av_clip(s->ir_channel, 0, s->channels - 1);
for (i = 0; i < s->w; i++) {
const double *b = s->iir[ch].ab[0];
const double *a = s->iir[ch].ab[1];
const int nb_b = s->iir[ch].nb_ab[0];
const int nb_a = s->iir[ch].nb_ab[1];
double w = i * M_PI / (s->w - 1);
double m, p;
get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p);
mag[i] = s->iir[ch].g * m;
phase[i] = p;
min = fmin(min, mag[i]);
max = fmax(max, mag[i]);
}
temp[0] = 0.;
for (i = 0; i < s->w - 1; i++) {
double d = phase[i] - phase[i + 1];
temp[i + 1] = ceil(fabs(d) / (2. * M_PI)) * 2. * M_PI * ((d > M_PI) - (d < -M_PI));
}
min_phase = phase[0];
max_phase = phase[0];
for (i = 1; i < s->w; i++) {
temp[i] += temp[i - 1];
phase[i] += temp[i];
min_phase = fmin(min_phase, phase[i]);
max_phase = fmax(max_phase, phase[i]);
}
for (i = 0; i < s->w - 1; i++) {
double div = s->w / (double)sample_rate;
delay[i + 1] = -(phase[i] - phase[i + 1]) / div;
min_delay = fmin(min_delay, delay[i + 1]);
max_delay = fmax(max_delay, delay[i + 1]);
}
delay[0] = delay[1];
for (i = 0; i < s->w; i++) {
int ymag = mag[i] / max * (s->h - 1);
int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
int yphase = (phase[i] - min_phase) / (max_phase - min_phase) * (s->h - 1);
ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
if (prev_ymag < 0)
prev_ymag = ymag;
if (prev_yphase < 0)
prev_yphase = yphase;
if (prev_ydelay < 0)
prev_ydelay = ydelay;
draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
prev_ymag = ymag;
prev_yphase = yphase;
prev_ydelay = ydelay;
}
if (s->w > 400 && s->h > 100) {
drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", max);
drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min);
drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
drawtext(out, 2, 22, "Max Phase:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", max_phase);
drawtext(out, 15 * 8 + 2, 22, text, 0xDDDDDDDD);
drawtext(out, 2, 32, "Min Phase:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min_phase);
drawtext(out, 15 * 8 + 2, 32, text, 0xDDDDDDDD);
drawtext(out, 2, 42, "Max Delay:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", max_delay);
drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD);
drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min_delay);
drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD);
}
end:
av_free(delay);
av_free(temp);
av_free(phase);
av_free(mag);
}
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->iir = av_calloc(s->channels, sizeof(*s->iir));
if (!s->iir)
return AVERROR(ENOMEM);
ret = read_gains(ctx, s->g_str, inlink->channels);
if (ret < 0)
return ret;
ret = read_channels(ctx, inlink->channels, s->a_str, 0);
if (ret < 0)
return ret;
ret = read_channels(ctx, inlink->channels, s->b_str, 1);
if (ret < 0)
return ret;
if (s->format == 2) {
convert_pr2zp(ctx, inlink->channels);
} else if (s->format == 3) {
convert_pd2zp(ctx, inlink->channels);
} else if (s->format == 4) {
convert_sp2zp(ctx, inlink->channels);
}
if (s->format > 0) {
check_stability(ctx, inlink->channels);
}
av_frame_free(&s->video);
if (s->response) {
s->video = ff_get_video_buffer(ctx->outputs[1], s->w, s->h);
if (!s->video)
return AVERROR(ENOMEM);
draw_response(ctx, s->video, inlink->sample_rate);
}
if (s->format == 0)
av_log(ctx, AV_LOG_WARNING, "tf coefficients format is not recommended for too high number of zeros/poles.\n");
if (s->format > 0 && s->process == 0) {
av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n");
ret = convert_zp2tf(ctx, inlink->channels);
if (ret < 0)
return ret;
} else if (s->format == 0 && s->process == 1) {
av_log(ctx, AV_LOG_ERROR, "Serial cascading is not implemented for transfer function.\n");
return AVERROR_PATCHWELCOME;
} else if (s->format > 0 && s->process == 1) {
if (inlink->format == AV_SAMPLE_FMT_S16P)
av_log(ctx, AV_LOG_WARNING, "Serial cascading is not recommended for i16 precision.\n");
ret = decompose_zp2biquads(ctx, inlink->channels);
if (ret < 0)
return ret;
}
for (ch = 0; s->format == 0 && ch < inlink->channels; ch++) {
IIRChannel *iir = &s->iir[ch];
for (i = 1; i < iir->nb_ab[0]; i++) {
iir->ab[0][i] /= iir->ab[0][0];
}
iir->ab[0][0] = 1.0;
for (i = 0; i < iir->nb_ab[1]; i++) {
iir->ab[1][i] *= iir->g;
}
normalize_coeffs(ctx, ch);
}
switch (inlink->format) {
case AV_SAMPLE_FMT_DBLP: s->iir_channel = s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break;
case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break;
case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break;
case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 1 ? iir_ch_serial_s16p : iir_ch_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];
ThreadData td;
AVFrame *out;
int ch, ret;
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);
}
td.in = in;
td.out = out;
ctx->internal->execute(ctx, s->iir_channel, &td, NULL, outlink->channels);
for (ch = 0; ch < outlink->channels; ch++) {
if (s->iir[ch].clippings > 0)
av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n",
ch, s->iir[ch].clippings);
s->iir[ch].clippings = 0;
}
if (in != out)
av_frame_free(&in);
if (s->response) {
AVFilterLink *outlink = ctx->outputs[1];
int64_t old_pts = s->video->pts;
int64_t new_pts = av_rescale_q(out->pts, ctx->inputs[0]->time_base, outlink->time_base);
if (new_pts > old_pts) {
AVFrame *clone;
s->video->pts = new_pts;
clone = av_frame_clone(s->video);
if (!clone)
return AVERROR(ENOMEM);
ret = ff_filter_frame(outlink, clone);
if (ret < 0)
return ret;
}
}
return ff_filter_frame(outlink, out);
}
static int config_video(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioIIRContext *s = ctx->priv;
outlink->sample_aspect_ratio = (AVRational){1,1};
outlink->w = s->w;
outlink->h = s->h;
outlink->frame_rate = s->rate;
outlink->time_base = av_inv_q(outlink->frame_rate);
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioIIRContext *s = ctx->priv;
AVFilterPad pad, vpad;
int ret;
if (!s->a_str || !s->b_str || !s->g_str) {
av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n");
return AVERROR(EINVAL);
}
switch (s->precision) {
case 0: s->sample_format = AV_SAMPLE_FMT_DBLP; break;
case 1: s->sample_format = AV_SAMPLE_FMT_FLTP; break;
case 2: s->sample_format = AV_SAMPLE_FMT_S32P; break;
case 3: s->sample_format = AV_SAMPLE_FMT_S16P; break;
default: return AVERROR_BUG;
}
pad = (AVFilterPad){
.name = av_strdup("default"),
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
};
if (!pad.name)
return AVERROR(ENOMEM);
ret = ff_insert_outpad(ctx, 0, &pad);
if (ret < 0)
return ret;
if (s->response) {
vpad = (AVFilterPad){
.name = av_strdup("filter_response"),
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video,
};
if (!vpad.name)
return AVERROR(ENOMEM);
ret = ff_insert_outpad(ctx, 1, &vpad);
if (ret < 0)
return ret;
}
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioIIRContext *s = ctx->priv;
int ch;
if (s->iir) {
for (ch = 0; ch < s->channels; ch++) {
IIRChannel *iir = &s->iir[ch];
av_freep(&iir->ab[0]);
av_freep(&iir->ab[1]);
av_freep(&iir->cache[0]);
av_freep(&iir->cache[1]);
av_freep(&iir->biquads);
}
}
av_freep(&s->iir);
av_freep(&ctx->output_pads[0].name);
if (s->response)
av_freep(&ctx->output_pads[1].name);
av_frame_free(&s->video);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
#define OFFSET(x) offsetof(AudioIIRContext, x)
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption aiir_options[] = {
{ "zeros", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
{ "z", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
{ "poles", "set A/denominator/poles coefficients", OFFSET(a_str),AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
{ "p", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
{ "gains", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
{ "k", "set channels gains", OFFSET(g_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 },
{ "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, AF, "format" },
{ "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, AF, "format" },
{ "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "format" },
{ "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "format" },
{ "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "format" },
{ "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "format" },
{ "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "format" },
{ "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "process" },
{ "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "process" },
{ "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "process" },
{ "s", "serial cascading", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" },
{ "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" },
{ "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" },
{ "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
{ "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
{ "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
{ "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision" },
{ "normalize", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
{ "n", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
{ "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
{ "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
{ "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
{ "rate", "set video rate", OFFSET(rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
{ 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),
.priv_class = &aiir_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = inputs,
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SLICE_THREADS,
};
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