/* * 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/avstring.h" #include "libavutil/intreadwrite.h" #include "libavutil/mem.h" #include "libavutil/opt.h" #include "libavutil/xga_font_data.h" #include "audio.h" #include "avfilter.h" #include "formats.h" #include "internal.h" #include "video.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 w1, w2; } BiquadContext; typedef struct IIRChannel { int nb_ab[2]; double *ab[2]; double g; double *cache[2]; double fir; 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; 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->incfg.formats)) < 0) return ret; } ret = ff_set_common_all_channel_counts(ctx); if (ret < 0) return ret; sample_fmts[0] = s->sample_format; ret = ff_set_common_formats_from_list(ctx, sample_fmts); if (ret < 0) return ret; return ff_set_common_all_samplerates(ctx); } #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; \ const double imix = 1. - 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 = nb_biquads - 1; i >= 0; 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 w1 = iir->biquads[i].w1; \ double w2 = iir->biquads[i].w2; \ \ for (n = 0; n < in->nb_samples; n++) { \ double i0 = ig * (i ? dst[n] : src[n]); \ double o0 = i0 * b0 + w1; \ \ w1 = b1 * i0 + w2 + a1 * o0; \ w2 = b2 * i0 + a2 * o0; \ o0 *= og * g; \ \ o0 = o0 * mix + imix * i0; \ 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].w1 = w1; \ iir->biquads[i].w2 = w2; \ } \ \ 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) #define PARALLEL_IIR_CH(name, type, min, max, need_clipping) \ static int iir_ch_parallel_## 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; \ const double imix = 1. - 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; \ const double fir = iir->fir; \ 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 b1 = iir->biquads[i].b[1]; \ const double b2 = iir->biquads[i].b[2]; \ double w1 = iir->biquads[i].w1; \ double w2 = iir->biquads[i].w2; \ \ for (n = 0; n < in->nb_samples; n++) { \ double i0 = ig * src[n]; \ double o0 = w1; \ \ w1 = b1 * i0 + w2 + a1 * o0; \ w2 = b2 * i0 + a2 * o0; \ o0 *= og * g; \ o0 += dst[n]; \ \ 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].w1 = w1; \ iir->biquads[i].w2 = w2; \ } \ \ for (n = 0; n < in->nb_samples; n++) { \ dst[n] += fir * src[n]; \ dst[n] = dst[n] * mix + imix * src[n]; \ } \ \ return 0; \ } PARALLEL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) PARALLEL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) PARALLEL_IIR_CH(fltp, float, -1., 1., 0) PARALLEL_IIR_CH(dblp, double, -1., 1., 0) #define LATTICE_IIR_CH(name, type, min, max, need_clipping) \ static int iir_ch_lattice_## 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 n0, n1, p0, *x = (double *)s->iir[ch].cache[0]; \ const int nb_stages = s->iir[ch].nb_ab[1]; \ const double *v = s->iir[ch].ab[0]; \ const double *k = 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++) { \ const double in = src[n] * ig; \ double out = 0.; \ \ n1 = in; \ for (int i = nb_stages - 1; i >= 0; i--) { \ n0 = n1 - k[i] * x[i]; \ p0 = n0 * k[i] + x[i]; \ out += p0 * v[i+1]; \ x[i] = p0; \ n1 = n0; \ } \ \ out += n1 * v[0]; \ memmove(&x[1], &x[0], nb_stages * sizeof(*x)); \ x[0] = n1; \ out *= og * g; \ out = out * mix + in * (1. - mix); \ if (need_clipping && out < min) { \ (*clippings)++; \ dst[n] = min; \ } else if (need_clipping && out > max) { \ (*clippings)++; \ dst[n] = max; \ } else { \ dst[n] = out; \ } \ } \ \ return 0; \ } LATTICE_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) LATTICE_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) LATTICE_IIR_CH(fltp, float, -1., 1., 0) LATTICE_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 (av_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 (av_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 (av_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 *const 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 > 0) { 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 biquad_process(double *x, double *y, int length, double b0, double b1, double b2, double a1, double a2) { double w1 = 0., w2 = 0.; a1 = -a1; a2 = -a2; for (int n = 0; n < length; n++) { double out, in = x[n]; y[n] = out = in * b0 + w1; w1 = b1 * in + w2 + a1 * out; w2 = b2 * in + a2 * out; } } static void solve(double *matrix, double *vector, int n, double *y, double *x, double *lu) { double sum = 0.; for (int i = 0; i < n; i++) { for (int j = i; j < n; j++) { sum = 0.; for (int k = 0; k < i; k++) sum += lu[i * n + k] * lu[k * n + j]; lu[i * n + j] = matrix[j * n + i] - sum; } for (int j = i + 1; j < n; j++) { sum = 0.; for (int k = 0; k < i; k++) sum += lu[j * n + k] * lu[k * n + i]; lu[j * n + i] = (1. / lu[i * n + i]) * (matrix[i * n + j] - sum); } } for (int i = 0; i < n; i++) { sum = 0.; for (int k = 0; k < i; k++) sum += lu[i * n + k] * y[k]; y[i] = vector[i] - sum; } for (int i = n - 1; i >= 0; i--) { sum = 0.; for (int k = i + 1; k < n; k++) sum += lu[i * n + k] * x[k]; x[i] = (1 / lu[i * n + i]) * (y[i] - sum); } } static int convert_serial2parallel(AVFilterContext *ctx, int channels) { AudioIIRContext *s = ctx->priv; for (int 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 length = nb_biquads * 2 + 1; double *impulse = av_calloc(length, sizeof(*impulse)); double *y = av_calloc(length, sizeof(*y)); double *resp = av_calloc(length, sizeof(*resp)); double *M = av_calloc((length - 1) * nb_biquads, 2 * 2 * sizeof(*M)); double *W; if (!impulse || !y || !resp || !M) { av_free(impulse); av_free(y); av_free(resp); av_free(M); return AVERROR(ENOMEM); } W = M + (length - 1) * 2 * nb_biquads; impulse[0] = 1.; for (int n = 0; n < nb_biquads; n++) { BiquadContext *biquad = &iir->biquads[n]; biquad_process(n ? y : impulse, y, length, biquad->b[0], biquad->b[1], biquad->b[2], biquad->a[1], biquad->a[2]); } for (int n = 0; n < nb_biquads; n++) { BiquadContext *biquad = &iir->biquads[n]; biquad_process(impulse, resp, length - 1, 1., 0., 0., biquad->a[1], biquad->a[2]); memcpy(M + n * 2 * (length - 1), resp, sizeof(*resp) * (length - 1)); memcpy(M + n * 2 * (length - 1) + length, resp, sizeof(*resp) * (length - 2)); memset(resp, 0, length * sizeof(*resp)); } solve(M, &y[1], length - 1, &impulse[1], resp, W); iir->fir = y[0]; for (int n = 0; n < nb_biquads; n++) { BiquadContext *biquad = &iir->biquads[n]; biquad->b[0] = 0.; biquad->b[1] = resp[n * 2 + 0]; biquad->b[2] = resp[n * 2 + 1]; } av_free(impulse); av_free(y); av_free(resp); av_free(M); } 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]; iir->ab[0][2*n] = exp(sr) * cos(si); iir->ab[0][2*n+1] = exp(sr) * sin(si); } for (n = 0; n < iir->nb_ab[1]; n++) { double sr = iir->ab[1][2*n]; double si = iir->ab[1][2*n+1]; iir->ab[1][2*n] = exp(sr) * cos(si); iir->ab[1][2*n+1] = exp(sr) * sin(si); } } } static double fact(double i) { if (i <= 0.) return 1.; return i * fact(i - 1.); } static double coef_sf2zf(double *a, int N, int n) { double z = 0.; for (int i = 0; i <= N; i++) { double acc = 0.; for (int k = FFMAX(n - N + i, 0); k <= FFMIN(i, n); k++) { acc += ((fact(i) * fact(N - i)) / (fact(k) * fact(i - k) * fact(n - k) * fact(N - i - n + k))) * ((k & 1) ? -1. : 1.); } z += a[i] * pow(2., i) * acc; } return z; } static void convert_sf2tf(AVFilterContext *ctx, int channels) { AudioIIRContext *s = ctx->priv; int ch; for (ch = 0; ch < channels; ch++) { IIRChannel *iir = &s->iir[ch]; double *temp0 = av_calloc(iir->nb_ab[0], sizeof(*temp0)); double *temp1 = av_calloc(iir->nb_ab[1], sizeof(*temp1)); if (!temp0 || !temp1) goto next; memcpy(temp0, iir->ab[0], iir->nb_ab[0] * sizeof(*temp0)); memcpy(temp1, iir->ab[1], iir->nb_ab[1] * sizeof(*temp1)); for (int n = 0; n < iir->nb_ab[0]; n++) iir->ab[0][n] = coef_sf2zf(temp0, iir->nb_ab[0] - 1, n); for (int n = 0; n < iir->nb_ab[1]; n++) iir->ab[1][n] = coef_sf2zf(temp1, iir->nb_ab[1] - 1, n); next: av_free(temp0); av_free(temp1); } } 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->ch_layout.nb_channels; s->iir = av_calloc(s->channels, sizeof(*s->iir)); if (!s->iir) return AVERROR(ENOMEM); ret = read_gains(ctx, s->g_str, inlink->ch_layout.nb_channels); if (ret < 0) return ret; ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->a_str, 0); if (ret < 0) return ret; ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->b_str, 1); if (ret < 0) return ret; if (s->format == -1) { convert_sf2tf(ctx, inlink->ch_layout.nb_channels); s->format = 0; } else if (s->format == 2) { convert_pr2zp(ctx, inlink->ch_layout.nb_channels); } else if (s->format == 3) { convert_pd2zp(ctx, inlink->ch_layout.nb_channels); } else if (s->format == 4) { convert_sp2zp(ctx, inlink->ch_layout.nb_channels); } if (s->format > 0) { check_stability(ctx, inlink->ch_layout.nb_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, "transfer function 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 processing is not recommended for zp coefficients format.\n"); ret = convert_zp2tf(ctx, inlink->ch_layout.nb_channels); if (ret < 0) return ret; } else if (s->format == -2 && s->process > 0) { av_log(ctx, AV_LOG_ERROR, "Only direct processing is implemented for lattice-ladder function.\n"); return AVERROR_PATCHWELCOME; } else if (s->format <= 0 && s->process == 1) { av_log(ctx, AV_LOG_ERROR, "Serial processing is not implemented for transfer function.\n"); return AVERROR_PATCHWELCOME; } else if (s->format <= 0 && s->process == 2) { av_log(ctx, AV_LOG_ERROR, "Parallel processing is not implemented for transfer function.\n"); return AVERROR_PATCHWELCOME; } else if (s->format > 0 && s->process == 1) { ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels); if (ret < 0) return ret; } else if (s->format > 0 && s->process == 2) { if (s->precision > 1) av_log(ctx, AV_LOG_WARNING, "Parallel processing is not recommended for fixed-point precisions.\n"); ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels); if (ret < 0) return ret; ret = convert_serial2parallel(ctx, inlink->ch_layout.nb_channels); if (ret < 0) return ret; } for (ch = 0; s->format == -2 && ch < inlink->ch_layout.nb_channels; ch++) { IIRChannel *iir = &s->iir[ch]; if (iir->nb_ab[0] != iir->nb_ab[1] + 1) { av_log(ctx, AV_LOG_ERROR, "Number of ladder coefficients must be one more than number of reflection coefficients.\n"); return AVERROR(EINVAL); } } for (ch = 0; s->format == 0 && ch < inlink->ch_layout.nb_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 == 2 ? iir_ch_parallel_dblp : s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break; case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 2 ? iir_ch_parallel_fltp : s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break; case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s32p : s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break; case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s16p : s->process == 1 ? iir_ch_serial_s16p : iir_ch_s16p; break; } if (s->format == -2) { switch (inlink->format) { case AV_SAMPLE_FMT_DBLP: s->iir_channel = iir_ch_lattice_dblp; break; case AV_SAMPLE_FMT_FLTP: s->iir_channel = iir_ch_lattice_fltp; break; case AV_SAMPLE_FMT_S32P: s->iir_channel = iir_ch_lattice_s32p; break; case AV_SAMPLE_FMT_S16P: s->iir_channel = iir_ch_lattice_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) && s->process != 2) { 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; ff_filter_execute(ctx, s->iir_channel, &td, NULL, outlink->ch_layout.nb_channels); for (ch = 0; ch < outlink->ch_layout.nb_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_append_outpad(ctx, &pad); if (ret < 0) return ret; if (s->response) { vpad = (AVFilterPad){ .name = "filter_response", .type = AVMEDIA_TYPE_VIDEO, .config_props = config_video, }; ret = ff_append_outpad(ctx, &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, }, }; #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/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, { "z", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, { "poles", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, { "p", "set A/denominator/poles/ladder 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}, -2, 4, AF, .unit = "format" }, { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, .unit = "format" }, { "ll", "lattice-ladder function", 0, AV_OPT_TYPE_CONST, {.i64=-2}, 0, 0, AF, .unit = "format" }, { "sf", "analog transfer function", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, .unit = "format" }, { "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "format" }, { "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "format" }, { "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "format" }, { "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "format" }, { "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, .unit = "format" }, { "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, .unit = "process" }, { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, .unit = "process" }, { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "process" }, { "s", "serial", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "process" }, { "p", "parallel", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "process" }, { "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, .unit = "precision" }, { "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, .unit = "precision" }, { "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" }, { "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" }, { "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" }, { "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "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); const 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, FILTER_INPUTS(inputs), FILTER_QUERY_FUNC(query_formats), .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS | AVFILTER_FLAG_SLICE_THREADS, };