/* * 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, };