/* * Copyright (c) 2020 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 "libavutil/cpu.h"#include "libavutil/channel_layout.h"#include "libavutil/ffmath.h"#include "libavutil/eval.h"#include "libavutil/mem.h"#include "libavutil/opt.h"#include "libavutil/tx.h"#include "audio.h"#include "avfilter.h"#include "filters.h"#include "formats.h"#include "window_func.h" typedef struct AudioFIRSourceContext {    const AVClass *class;     char *freq_points_str;    char *magnitude_str;    char *phase_str;    int nb_taps;    int sample_rate;    int nb_samples;    int win_func;    int preset;    int interp;    int phaset;     AVComplexFloat *complexf;    float *freq;    float *magnitude;    float *phase;    int freq_size;    int magnitude_size;    int phase_size;    int nb_freq;    int nb_magnitude;    int nb_phase;     float *taps;    float *win;    int64_t pts;     AVTXContext *tx_ctx, *itx_ctx;    av_tx_fn tx_fn, itx_fn;} AudioFIRSourceContext; #define OFFSET(x) offsetof(AudioFIRSourceContext, x)#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption afirsrc_options[] = {    { "taps",      "set number of taps",   OFFSET(nb_taps),         AV_OPT_TYPE_INT,    {.i64=1025}, 9, UINT16_MAX, FLAGS },    { "t",         "set number of taps",   OFFSET(nb_taps),         AV_OPT_TYPE_INT,    {.i64=1025}, 9, UINT16_MAX, FLAGS },    { "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },    { "f",         "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },    { "magnitude", "set magnitude values", OFFSET(magnitude_str),   AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },    { "m",         "set magnitude values", OFFSET(magnitude_str),   AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },    { "phase",     "set phase values",     OFFSET(phase_str),       AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },    { "p",         "set phase values",     OFFSET(phase_str),       AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },    { "sample_rate", "set sample rate",    OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100},  1, INT_MAX,    FLAGS },    { "r",           "set sample rate",    OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100},  1, INT_MAX,    FLAGS },    { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },    { "n",          "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },    WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),    WIN_FUNC_OPTION("w",        OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),    {NULL}}; AVFILTER_DEFINE_CLASS(afirsrc); static av_cold int init(AVFilterContext *ctx){    AudioFIRSourceContext *s = ctx->priv;     if (!(s->nb_taps & 1)) {        av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps);        s->nb_taps |= 1;    }     return 0;} static av_cold void uninit(AVFilterContext *ctx){    AudioFIRSourceContext *s = ctx->priv;     av_freep(&s->win);    av_freep(&s->taps);    av_freep(&s->freq);    av_freep(&s->magnitude);    av_freep(&s->phase);    av_freep(&s->complexf);    av_tx_uninit(&s->tx_ctx);    av_tx_uninit(&s->itx_ctx);} static av_cold int query_formats(const AVFilterContext *ctx,                                 AVFilterFormatsConfig **cfg_in,                                 AVFilterFormatsConfig **cfg_out){    const AudioFIRSourceContext *s = ctx->priv;    static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };    int sample_rates[] = { s->sample_rate, -1 };    static const enum AVSampleFormat sample_fmts[] = {        AV_SAMPLE_FMT_FLT,        AV_SAMPLE_FMT_NONE    };    int ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts);    if (ret < 0)        return ret;     ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, chlayouts);    if (ret < 0)        return ret;     return ff_set_common_samplerates_from_list2(ctx, cfg_in, cfg_out, sample_rates);} static int parse_string(char *str, float **items, int *nb_items, int *items_size){    float *new_items;    char *tail;     new_items = av_fast_realloc(NULL, items_size, sizeof(float));    if (!new_items)        return AVERROR(ENOMEM);    *items = new_items;     tail = str;    if (!tail)        return AVERROR(EINVAL);     do {        (*items)[(*nb_items)++] = av_strtod(tail, &tail);        new_items = av_fast_realloc(*items, items_size, (*nb_items + 2) * sizeof(float));        if (!new_items)            return AVERROR(ENOMEM);        *items = new_items;        if (tail && *tail)            tail++;    } while (tail && *tail);     return 0;} static void lininterp(AVComplexFloat *complexf,                      const float *freq,                      const float *magnitude,                      const float *phase,                      int m, int minterp){    for (int i = 0; i < minterp; i++) {        for (int j = 1; j < m; j++) {            const float x = i / (float)minterp;             if (x <= freq[j]) {                const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];                const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];                 complexf[i].re = mg * cosf(ph);                complexf[i].im = mg * sinf(ph);                break;            }        }    }} static av_cold int config_output(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    AudioFIRSourceContext *s = ctx->priv;    float overlap, scale = 1.f, compensation;    int fft_size, middle, ret;     s->nb_freq = s->nb_magnitude = s->nb_phase = 0;     ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);    if (ret < 0)        return ret;     ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);    if (ret < 0)        return ret;     ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size);    if (ret < 0)        return ret;     if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) {        av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n");        return AVERROR(EINVAL);    }     for (int i = 0; i < s->nb_freq; i++) {        if (i == 0 && s->freq[i] != 0.f) {            av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n");            return AVERROR(EINVAL);        }         if (i == s->nb_freq - 1 && s->freq[i] != 1.f) {            av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n");            return AVERROR(EINVAL);        }         if (i && s->freq[i] < s->freq[i-1]) {            av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n");            return AVERROR(EINVAL);        }    }     fft_size = 1 << (av_log2(s->nb_taps) + 1);    s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf));    if (!s->complexf)        return AVERROR(ENOMEM);     ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);    if (ret < 0)        return ret;     s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));    if (!s->taps)        return AVERROR(ENOMEM);     s->win = av_calloc(s->nb_taps, sizeof(*s->win));    if (!s->win)        return AVERROR(ENOMEM);     generate_window_func(s->win, s->nb_taps, s->win_func, &overlap);     lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2);     s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(*s->complexf));     compensation = 2.f / fft_size;    middle = s->nb_taps / 2;     for (int i = 0; i <= middle; i++) {        s->taps[         i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i];        s->taps[middle + i] = s->complexf[fft_size          + i].re * compensation * s->win[middle + i];    }     s->pts = 0;     return 0;} static int activate(AVFilterContext *ctx){    AVFilterLink *outlink = ctx->outputs[0];    AudioFIRSourceContext *s = ctx->priv;    AVFrame *frame;    int nb_samples;     if (!ff_outlink_frame_wanted(outlink))        return FFERROR_NOT_READY;     nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);    if (nb_samples <= 0) {        ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);        return 0;    }     if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))        return AVERROR(ENOMEM);     memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));     frame->pts = s->pts;    s->pts    += nb_samples;    return ff_filter_frame(outlink, frame);} static const AVFilterPad afirsrc_outputs[] = {    {        .name          = "default",        .type          = AVMEDIA_TYPE_AUDIO,        .config_props  = config_output,    },}; const FFFilter ff_asrc_afirsrc = {    .p.name        = "afirsrc",    .p.description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."),    .p.priv_class  = &afirsrc_class,    .init          = init,    .uninit        = uninit,    .activate      = activate,    .priv_size     = sizeof(AudioFIRSourceContext),    FILTER_OUTPUTS(afirsrc_outputs),    FILTER_QUERY_FUNC2(query_formats),}; #define DEFAULT_BANDS "25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000" typedef struct EqPreset {    char name[16];    float gains[16];} EqPreset; static const EqPreset eq_presets[] = {    { "flat",          { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },    { "acoustic",      { 5.0, 4.5, 4.0, 3.5, 1.5, 1.0, 1.5, 1.5, 2.0, 3.0, 3.5, 4.0, 3.7, 3.0, 3.0 } },    { "bass",          { 10.0, 8.8, 8.5, 6.5, 2.5, 1.5, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },    { "beats",         { -5.5, -5.0, -4.5, -4.2, -3.5, -3.0, -1.9, 0, 0, 0, 0, 0, 0, 0, 0 } },    { "classic",       { -0.3, 0.3, -3.5, -9.0, -1.0, 0.0, 1.8, 2.1, 0.0, 0.0, 0.0, 4.4, 9.0, 9.0, 9.0 } },    { "clear",         { 3.5, 5.5, 6.5, 9.5, 8.0, 6.5, 3.5, 2.5, 1.3, 5.0, 7.0, 9.0, 10.0, 11.0, 9.0 } },    { "deep bass",     { 12.0, 8.0, 0.0, -6.7, -12.0, -9.0, -3.5, -3.5, -6.1, 0.0, -3.0, -5.0, 0.0, 1.2, 3.0 } },    { "dubstep",       { 12.0, 10.0, 0.5, -1.0, -3.0, -5.0, -5.0, -4.8, -4.5, -2.5, -1.0, 0.0, -2.5, -2.5, 0.0 } },    { "electronic",    { 4.0, 4.0, 3.5, 1.0, 0.0, -0.5, -2.0, 0.0, 2.0, 0.0, 0.0, 1.0, 3.0, 4.0, 4.5 } },    { "hardstyle",     { 6.1, 7.0, 12.0, 6.1, -5.0, -12.0, -2.5, 3.0, 6.5, 0.0, -2.2, -4.5, -6.1, -9.2, -10.0 } },    { "hip-hop",       { 4.5, 4.3, 4.0, 2.5, 1.5, 3.0, -1.0, -1.5, -1.5, 1.5, 0.0, -1.0, 0.0, 1.5, 3.0 } },    { "jazz",          { 0.0, 0.0, 0.0, 2.0, 4.0, 5.9, -5.9, -4.5, -2.5, 2.5, 1.0, -0.8, -0.8, -0.8, -0.8 } },    { "metal",         { 10.5, 10.5, 7.5, 0.0, 2.0, 5.5, 0.0, 0.0, 0.0, 6.1, 0.0, 0.0, 6.1, 10.0, 12.0 } },    { "movie",         { 3.0, 3.0, 6.1, 8.5, 9.0, 7.0, 6.1, 6.1, 5.0, 8.0, 3.5, 3.5, 8.0, 10.0, 8.0 } },    { "pop",           { 0.0, 0.0, 0.0, 0.0, 0.0, 1.3, 2.0, 2.5, 5.0, -1.5, -2.0, -3.0, -3.0, -3.0, -3.0 } },    { "r&b",           { 3.0, 3.0, 7.0, 6.1, 4.5, 1.5, -1.5, -2.0, -1.5, 2.0, 2.5, 3.0, 3.5, 3.8, 4.0 } },    { "rock",          { 0.0, 0.0, 0.0, 3.0, 3.0, -10.0, -4.0, -1.0, 0.8, 3.0, 3.0, 3.0, 3.0, 3.0, 3.0 } },    { "vocal booster", { -1.5, -2.0, -3.0, -3.0, -0.5, 1.5, 3.5, 3.5, 3.5, 3.0, 2.0, 1.5, 0.0, 0.0, -1.5 } },}; static const AVOption afireqsrc_options[] = {    { "preset","set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" },    { "p",     "set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" },    { "custom",            NULL, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 0].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 0}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 1].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 1}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 2].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 2}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 3].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 3}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 4].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 4}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 5].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 5}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 6].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 6}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 7].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 7}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 8].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 8}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[ 9].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 9}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[10].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[11].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[12].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[13].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[14].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[15].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[16].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, FLAGS, .unit = "preset" },    { eq_presets[17].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=17}, 0, 0, FLAGS, .unit = "preset" },    { "gains", "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS },    { "g",     "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS },    { "bands", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },    { "b",     "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },    { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },    { "t",    "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },    { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100},  1, INT_MAX,    FLAGS },    { "r",           "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100},  1, INT_MAX,    FLAGS },    { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },    { "n",          "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },    { "interp","set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" },    { "i",     "set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" },    { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "interp" },    { "cubic",  NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "interp" },    { "phase","set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" },    { "h",    "set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" },    { "linear", "linear phase",  0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "phase" },    { "min",    "minimum phase", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "phase" },    {NULL}}; AVFILTER_DEFINE_CLASS(afireqsrc); static void eq_interp(AVComplexFloat *complexf,                      const float *freq,                      const float *magnitude,                      int m, int interp, int minterp,                      const float factor){    for (int i = 0; i < minterp; i++) {        for (int j = 0; j < m; j++) {            const float x = factor * i;             if (x <= freq[j+1]) {                float g;                 if (interp == 0) {                    const float d  = freq[j+1] - freq[j];                    const float d0 = x - freq[j];                    const float d1 = freq[j+1] - x;                    const float g0 = magnitude[j];                    const float g1 = magnitude[j+1];                     if (d0 && d1) {                        g = (d0 * g1 + d1 * g0) / d;                    } else if (d0) {                        g = g1;                    } else {                        g = g0;                    }                } else {                    if (x <= freq[j]) {                        g = magnitude[j];                    } else {                        float x1, x2, x3;                        float a, b, c, d;                        float m0, m1, m2, msum;                        const float unit = freq[j+1] - freq[j];                         m0 = j != 0 ? unit * (magnitude[j] - magnitude[j-1]) / (freq[j] - freq[j-1]) : 0;                        m1 = magnitude[j+1] - magnitude[j];                        m2 = j != minterp - 1 ? unit * (magnitude[j+2] - magnitude[j+1]) / (freq[j+2] - freq[j+1]) : 0;                         msum = fabsf(m0) + fabsf(m1);                        m0 = msum > 0.f ? (fabsf(m0) * m1 + fabsf(m1) * m0) / msum : 0.f;                        msum = fabsf(m1) + fabsf(m2);                        m1 = msum > 0.f ? (fabsf(m1) * m2 + fabsf(m2) * m1) / msum : 0.f;                         d = magnitude[j];                        c = m0;                        b = 3.f * magnitude[j+1] - m1 - 2.f * c - 3.f * d;                        a = magnitude[j+1] - b - c - d;                         x1 = (x - freq[j]) / unit;                        x2 = x1 * x1;                        x3 = x2 * x1;                         g = a * x3 + b * x2 + c * x1 + d;                    }                }                 complexf[i].re = g;                complexf[i].im = 0;                complexf[minterp * 2 - i - 1].re = g;                complexf[minterp * 2 - i - 1].im = 0;                 break;            }        }    }} static av_cold int config_eq_output(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    AudioFIRSourceContext *s = ctx->priv;    int fft_size, middle, asize, ret;    float scale, factor;     s->nb_freq = s->nb_magnitude = 0;    if (s->preset < 0) {        ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);        if (ret < 0)            return ret;         ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);        if (ret < 0)            return ret;    } else {        char *freq_str;         s->nb_magnitude = FF_ARRAY_ELEMS(eq_presets[s->preset].gains);         freq_str = av_strdup(DEFAULT_BANDS);        if (!freq_str)            return AVERROR(ENOMEM);         ret = parse_string(freq_str, &s->freq, &s->nb_freq, &s->freq_size);        av_free(freq_str);        if (ret < 0)            return ret;         s->magnitude = av_calloc(s->nb_magnitude + 1, sizeof(*s->magnitude));        if (!s->magnitude)            return AVERROR(ENOMEM);        memcpy(s->magnitude, eq_presets[s->preset].gains, sizeof(*s->magnitude) * s->nb_magnitude);    }     if (s->nb_freq != s->nb_magnitude || s->nb_freq < 2) {        av_log(ctx, AV_LOG_ERROR, "Number of bands and gains must be same and >= 2.\n");        return AVERROR(EINVAL);    }     s->freq[s->nb_freq] = outlink->sample_rate * 0.5f;    s->magnitude[s->nb_freq] = s->magnitude[s->nb_freq-1];     fft_size = s->nb_taps * 2;    factor = FFMIN(outlink->sample_rate * 0.5f, s->freq[s->nb_freq - 1]) / (float)fft_size;    asize = FFALIGN(fft_size, av_cpu_max_align());

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