/*
* 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/opt.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 a0, a1, a2;
double b0, b1, b2;
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;
int format;
int process;
int precision;
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
};
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;
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; \
ThreadData *td = arg; \
AVFrame *in = td->in, *out = td->out; \
const type *src = (const type *)in->extended_data[ch]; \
double *ic = (double *)s->iir[ch].cache[0]; \
double *oc = (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]; \
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; \
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; \
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]; \
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].a1; \
const double a2 = -iir->biquads[i].a2; \
const double b0 = iir->biquads[i].b0; \
const double b1 = iir->biquads[i].b1; \
const double b2 = iir->biquads[i].b2; \
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; \
\
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" };
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 multiply(double wre, double wim, int npz, double *coeffs)
{
double nwre = -wre, nwim = -wim;
double cre, cim;
int i;
for (i = npz; i >= 1; i--) {
cre = coeffs[2 * i + 0];
cim = coeffs[2 * i + 1];
coeffs[2 * i + 0] = (nwre * cre - nwim * cim) + coeffs[2 * (i - 1) + 0];
coeffs[2 * i + 1] = (nwre * cim + nwim * cre) + coeffs[2 * (i - 1) + 1];
}
cre = coeffs[0];
cim = coeffs[1];
coeffs[0] = nwre * cre - nwim * cim;
coeffs[1] = nwre * cim + nwim * cre;
}
static int expand(AVFilterContext *ctx, double *pz, int nb, double *coeffs)
{
int i;
coeffs[0] = 1.0;
coeffs[1] = 0.0;
for (i = 0; i < nb; i++) {
coeffs[2 * (i + 1) ] = 0.0;
coeffs[2 * (i + 1) + 1] = 0.0;
}
for (i = 0; i < nb; i++)
multiply(pz[2 * i], pz[2 * i + 1], nb, coeffs);
for (i = 0; i < nb + 1; i++) {
if (fabs(coeffs[2 * i + 1]) > FLT_EPSILON) {
av_log(ctx, AV_LOG_ERROR, "coeff: %lf of z^%d is not real; poles/zeros are not complex conjugates.\n",
coeffs[2 * i + 1], i);
return AVERROR(EINVAL);
}
}
return 0;
}
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[0] + 1) * 2, sizeof(*topc));
botc = av_calloc((iir->nb_ab[1] + 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]++;
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;
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[1]; 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].a0 = 1.0;
iir->biquads[current_biquad].a1 = a[2] / a[4];
iir->biquads[current_biquad].a2 = a[0] / a[4];
iir->biquads[current_biquad].b0 = b[4] / a[4] * (current_biquad ? 1.0 : iir->g);
iir->biquads[current_biquad].b1 = b[2] / a[4] * (current_biquad ? 1.0 : iir->g);
iir->biquads[current_biquad].b2 = b[0] / a[4] * (current_biquad ? 1.0 : iir->g);
av_log(ctx, AV_LOG_VERBOSE, "a=%lf %lf %lf:b=%lf %lf %lf\n",
iir->biquads[current_biquad].a0,
iir->biquads[current_biquad].a1,
iir->biquads[current_biquad].a2,
iir->biquads[current_biquad].b0,
iir->biquads[current_biquad].b1,
iir->biquads[current_biquad].b2);
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_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 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);
}
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; 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];
}
for (i = 0; i < iir->nb_ab[1]; i++) {
iir->ab[1][i] *= iir->g / iir->ab[0][0];
}
}
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;
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);
return ff_filter_frame(outlink, out);
}
static av_cold int init(AVFilterContext *ctx)
{
AudioIIRContext *s = ctx->priv;
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;
}
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);
}
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[] = {
{ "z", "set B/numerator/zeros coefficients", OFFSET(b_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 },
{ "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 },
{ "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 3, AF, "format" },
{ "tf", "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" },
{ "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" },
{ "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" },
{ 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,
.outputs = outputs,
.flags = AVFILTER_FLAG_SLICE_THREADS,
};