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FFmpeg/libavfilter/af_aiir.c
Andreas Rheinhardt 97b1a2c564 avfilter/af_aiir: Fix segfault and leak upon allocation failure 
The aiir filter adds output pads in its init function. Each of these
output pads had a name which was allocated and to be freed in the uninit
function. Given that the aiir filter has between one and two outputs,
one output pad's name was freed unconditionally and a second was freed
conditionally.

Yet if adding output pads fails, there are no output pads at all and
trying to free a nonexistent pad's name will lead to a segfault.

Furthermore, if the name could be successfully allocated, yet adding the
new pad fails, the name would leak.

This commit fixes this by not allocating the pads' names at all any
more: They are constant anyway. This allows to remove the code to free
them and hence fixes the aforementioned bugs.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-08-26 23:52:56 +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 = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
};
ret = ff_insert_outpad(ctx, 0, &pad);
if (ret < 0)
return ret;
if (s->response) {
vpad = (AVFilterPad){
.name = "filter_response",
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video,
};
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_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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