/* * 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 */ /** * @file * Crossover filter * * Split an audio stream into several bands. */ #include "libavutil/attributes.h"#include "libavutil/avstring.h"#include "libavutil/channel_layout.h"#include "libavutil/float_dsp.h"#include "libavutil/internal.h"#include "libavutil/mem.h"#include "libavutil/opt.h" #include "audio.h"#include "avfilter.h"#include "filters.h"#include "formats.h" #define MAX_SPLITS 16#define MAX_BANDS MAX_SPLITS + 1 #define B0 0#define B1 1#define B2 2#define A1 3#define A2 4 typedef struct BiquadCoeffs {    double cd[5];    float cf[5];} BiquadCoeffs; typedef struct AudioCrossoverContext {    const AVClass *class;     char *splits_str;    char *gains_str;    int order_opt;    float level_in;    int precision;     int order;    int filter_count;    int first_order;    int ap_filter_count;    int nb_splits;    float splits[MAX_SPLITS];     float gains[MAX_BANDS];     BiquadCoeffs lp[MAX_BANDS][20];    BiquadCoeffs hp[MAX_BANDS][20];    BiquadCoeffs ap[MAX_BANDS][20];     AVFrame *xover;     AVFrame *frames[MAX_BANDS];     int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);     AVFloatDSPContext *fdsp;} AudioCrossoverContext; #define OFFSET(x) offsetof(AudioCrossoverContext, x)#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM static const AVOption acrossover_options[] = {    { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },    { "order", "set filter order",      OFFSET(order_opt),  AV_OPT_TYPE_INT,    {.i64=1},     0, 9, AF, .unit = "m" },    { "2nd",   "2nd order (12 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0, 0, AF, .unit = "m" },    { "4th",   "4th order (24 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0, 0, AF, .unit = "m" },    { "6th",   "6th order (36 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=2},     0, 0, AF, .unit = "m" },    { "8th",   "8th order (48 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=3},     0, 0, AF, .unit = "m" },    { "10th",  "10th order (60 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=4},     0, 0, AF, .unit = "m" },    { "12th",  "12th order (72 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=5},     0, 0, AF, .unit = "m" },    { "14th",  "14th order (84 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=6},     0, 0, AF, .unit = "m" },    { "16th",  "16th order (96 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=7},     0, 0, AF, .unit = "m" },    { "18th",  "18th order (108 dB/8ve)",0,                 AV_OPT_TYPE_CONST,  {.i64=8},     0, 0, AF, .unit = "m" },    { "20th",  "20th order (120 dB/8ve)",0,                 AV_OPT_TYPE_CONST,  {.i64=9},     0, 0, AF, .unit = "m" },    { "level", "set input gain",        OFFSET(level_in),   AV_OPT_TYPE_FLOAT,  {.dbl=1},     0, 1, AF },    { "gain",  "set output bands gain", OFFSET(gains_str),  AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },    { "precision",  "set processing precision", OFFSET(precision),   AV_OPT_TYPE_INT,   {.i64=0}, 0, 2, AF, .unit = "precision" },    {  "auto",  "set auto processing precision",                  0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },    {  "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },    {  "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },    { NULL }}; AVFILTER_DEFINE_CLASS(acrossover); static int query_formats(const AVFilterContext *ctx,                         AVFilterFormatsConfig **cfg_in,                         AVFilterFormatsConfig **cfg_out){    const AudioCrossoverContext *s = ctx->priv;    static const enum AVSampleFormat auto_sample_fmts[] = {        AV_SAMPLE_FMT_FLTP,        AV_SAMPLE_FMT_DBLP,        AV_SAMPLE_FMT_NONE    };    enum AVSampleFormat sample_fmts[] = {        AV_SAMPLE_FMT_FLTP,        AV_SAMPLE_FMT_NONE    };    const enum AVSampleFormat *sample_fmts_list = sample_fmts;    int ret;     switch (s->precision) {    case 0:        sample_fmts_list = auto_sample_fmts;        break;    case 1:        sample_fmts[0] = AV_SAMPLE_FMT_FLTP;        break;    case 2:        sample_fmts[0] = AV_SAMPLE_FMT_DBLP;        break;    default:        break;    }    ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts_list);    if (ret < 0)        return ret;     return 0;} static int parse_gains(AVFilterContext *ctx){    AudioCrossoverContext *s = ctx->priv;    char *p, *arg, *saveptr = NULL;    int i, ret = 0;     saveptr = NULL;    p = s->gains_str;    for (i = 0; i < MAX_BANDS; i++) {        float gain;        char c[3] = { 0 };         if (!(arg = av_strtok(p, " |", &saveptr)))            break;         p = NULL;         if (av_sscanf(arg, "%f%2s", &gain, c) < 1) {            av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);            ret = AVERROR(EINVAL);            break;        }         if (c[0] == 'd' && c[1] == 'B')            s->gains[i] = expf(gain * M_LN10 / 20.f);        else            s->gains[i] = gain;    }     for (; i < MAX_BANDS; i++)        s->gains[i] = 1.f;     return ret;} static av_cold int init(AVFilterContext *ctx){    AudioCrossoverContext *s = ctx->priv;    char *p, *arg, *saveptr = NULL;    int i, ret = 0;     s->fdsp = avpriv_float_dsp_alloc(0);    if (!s->fdsp)        return AVERROR(ENOMEM);     p = s->splits_str;    for (i = 0; i < MAX_SPLITS; i++) {        float freq;         if (!(arg = av_strtok(p, " |", &saveptr)))            break;         p = NULL;         if (av_sscanf(arg, "%f", &freq) != 1) {            av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);            return AVERROR(EINVAL);        }        if (freq <= 0) {            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);            return AVERROR(EINVAL);        }         if (i > 0 && freq <= s->splits[i-1]) {            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);            return AVERROR(EINVAL);        }         s->splits[i] = freq;    }     s->nb_splits = i;     ret = parse_gains(ctx);    if (ret < 0)        return ret;     for (i = 0; i <= s->nb_splits; i++) {        AVFilterPad pad  = { 0 };        char *name;         pad.type = AVMEDIA_TYPE_AUDIO;        name = av_asprintf("out%d", ctx->nb_outputs);        if (!name)            return AVERROR(ENOMEM);        pad.name = name;         if ((ret = ff_append_outpad_free_name(ctx, &pad)) < 0)            return ret;    }     return ret;} static void set_lp(BiquadCoeffs *b, double fc, double q, double sr){    double omega = 2. * M_PI * fc / sr;    double cosine = cos(omega);    double alpha = sin(omega) / (2. * q);     double b0 = (1. - cosine) / 2.;    double b1 = 1. - cosine;    double b2 = (1. - cosine) / 2.;    double a0 = 1. + alpha;    double a1 = -2. * cosine;    double a2 = 1. - alpha;     b->cd[B0] =  b0 / a0;    b->cd[B1] =  b1 / a0;    b->cd[B2] =  b2 / a0;    b->cd[A1] = -a1 / a0;    b->cd[A2] = -a2 / a0;     b->cf[B0] = b->cd[B0];    b->cf[B1] = b->cd[B1];    b->cf[B2] = b->cd[B2];    b->cf[A1] = b->cd[A1];    b->cf[A2] = b->cd[A2];} static void set_hp(BiquadCoeffs *b, double fc, double q, double sr){    double omega = 2. * M_PI * fc / sr;    double cosine = cos(omega);    double alpha = sin(omega) / (2. * q);     double b0 = (1. + cosine) / 2.;    double b1 = -1. - cosine;    double b2 = (1. + cosine) / 2.;    double a0 = 1. + alpha;    double a1 = -2. * cosine;    double a2 = 1. - alpha;     b->cd[B0] =  b0 / a0;    b->cd[B1] =  b1 / a0;    b->cd[B2] =  b2 / a0;    b->cd[A1] = -a1 / a0;    b->cd[A2] = -a2 / a0;     b->cf[B0] = b->cd[B0];    b->cf[B1] = b->cd[B1];    b->cf[B2] = b->cd[B2];    b->cf[A1] = b->cd[A1];    b->cf[A2] = b->cd[A2];} static void set_ap(BiquadCoeffs *b, double fc, double q, double sr){    double omega = 2. * M_PI * fc / sr;    double cosine = cos(omega);    double alpha = sin(omega) / (2. * q);     double a0 = 1. + alpha;    double a1 = -2. * cosine;    double a2 = 1. - alpha;    double b0 = a2;    double b1 = a1;    double b2 = a0;     b->cd[B0] =  b0 / a0;    b->cd[B1] =  b1 / a0;    b->cd[B2] =  b2 / a0;    b->cd[A1] = -a1 / a0;    b->cd[A2] = -a2 / a0;     b->cf[B0] = b->cd[B0];    b->cf[B1] = b->cd[B1];    b->cf[B2] = b->cd[B2];    b->cf[A1] = b->cd[A1];    b->cf[A2] = b->cd[A2];} static void set_ap1(BiquadCoeffs *b, double fc, double sr){    double omega = 2. * M_PI * fc / sr;     b->cd[A1] = exp(-omega);    b->cd[A2] = 0.;    b->cd[B0] = -b->cd[A1];    b->cd[B1] = 1.;    b->cd[B2] = 0.;     b->cf[B0] = b->cd[B0];    b->cf[B1] = b->cd[B1];    b->cf[B2] = b->cd[B2];    b->cf[A1] = b->cd[A1];    b->cf[A2] = b->cd[A2];} static void calc_q_factors(int order, double *q){    double n = order / 2.;     for (int i = 0; i < n / 2; i++)        q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));} #define BIQUAD_PROCESS(name, type)                             \static void biquad_process_## name(const type *const c,        \                                   type *b,                    \                                   type *dst, const type *src, \                                   int nb_samples)             \{                                                              \    const type b0 = c[B0];                                     \    const type b1 = c[B1];                                     \    const type b2 = c[B2];                                     \    const type a1 = c[A1];                                     \    const type a2 = c[A2];                                     \    type z1 = b[0];                                            \    type z2 = b[1];                                            \                                                               \    for (int n = 0; n + 1 < nb_samples; n++) {                 \        type in = src[n];                                      \        type out;                                              \                                                               \        out = in * b0 + z1;                                    \        z1 = b1 * in + z2 + a1 * out;                          \        z2 = b2 * in + a2 * out;                               \        dst[n] = out;                                          \                                                               \        n++;                                                   \        in = src[n];                                           \        out = in * b0 + z1;                                    \        z1 = b1 * in + z2 + a1 * out;                          \        z2 = b2 * in + a2 * out;                               \        dst[n] = out;                                          \    }                                                          \                                                               \    if (nb_samples & 1) {                                      \        const int n = nb_samples - 1;                          \        const type in = src[n];                                \        type out;                                              \                                                               \        out = in * b0 + z1;                                    \        z1 = b1 * in + z2 + a1 * out;                          \        z2 = b2 * in + a2 * out;                               \        dst[n] = out;                                          \    }                                                          \                                                               \    b[0] = z1;                                                 \    b[1] = z2;                                                 \} BIQUAD_PROCESS(fltp, float)BIQUAD_PROCESS(dblp, double) #define XOVER_PROCESS(name, type, one, ff)                                                  \static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \{                                                                                           \    AudioCrossoverContext *s = ctx->priv;                                                   \    AVFrame *in = arg;                                                           \    AVFrame **frames = s->frames;                                                           \    const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;                        \    const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;                      \    const int nb_samples = in->nb_samples;                                                  \    const int nb_outs = ctx->nb_outputs;                                                    \    const int first_order = s->first_order;                                                 \                                                                                            \    for (int ch = start; ch < end; ch++) {                                                  \        const type *src = (const type *)in->extended_data[ch];                              \        type *xover = (type *)s->xover->extended_data[ch];                                  \                                                                                            \        s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src,       \                                    s->level_in, FFALIGN(nb_samples, sizeof(type)));        \                                                                                            \        for (int band = 0; band < nb_outs; band++) {                                        \            for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) {               \                const type *prv = (const type *)frames[band]->extended_data[ch];            \                type *dst = (type *)frames[band + 1]->extended_data[ch];                    \                const type *hsrc = f == 0 ? prv : dst;                                      \                type *hp = xover + nb_outs * 20 + band * 20 + f * 2;                        \                const type *const hpc = (type *)&s->hp[band][f].c ## ff;                    \                                                                                            \                biquad_process_## name(hpc, hp, dst, hsrc, nb_samples);                     \            }                                                                               \                                                                                            \            for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) {               \                type *dst = (type *)frames[band]->extended_data[ch];                        \                const type *lsrc = dst;                                                     \                type *lp = xover + band * 20 + f * 2;                                       \                const type *const lpc = (type *)&s->lp[band][f].c ## ff;                    \                                                                                            \                biquad_process_## name(lpc, lp, dst, lsrc, nb_samples);                     \            }                                                                               \                                                                                            \            for (int aband = band + 1; aband + 1 < nb_outs; aband++) {                      \                if (first_order) {                                                          \                    const type *asrc = (const type *)frames[band]->extended_data[ch];       \                    type *dst = (type *)frames[band]->extended_data[ch];                    \                    type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20;        \                    const type *const apc = (type *)&s->ap[aband][0].c ## ff;               \                                                                                            \                    biquad_process_## name(apc, ap, dst, asrc, nb_samples);                 \                }                                                                           \                                                                                            \                for (int f = first_order; f < s->ap_filter_count; f++) {                    \                    const type *asrc = (const type *)frames[band]->extended_data[ch];       \                    type *dst = (type *)frames[band]->extended_data[ch];                    \                    type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\                    const type *const apc = (type *)&s->ap[aband][f].c ## ff;               \                                                                                            \                    biquad_process_## name(apc, ap, dst, asrc, nb_samples);                 \                }                                                                           \            }                                                                               \        }                                                                                   \                                                                                            \        for (int band = 0; band < nb_outs; band++) {                                        \            const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one);    \            type *dst = (type *)frames[band]->extended_data[ch];                            \                                                                                            \            s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain,                              \                                               FFALIGN(nb_samples, sizeof(type)));          \        }                                                                                   \    }                                                                                       \                                                                                            \    return 0;                                                                               \} XOVER_PROCESS(fltp, float, 1.f, f)XOVER_PROCESS(dblp, double, 1.0, d) static int config_input(AVFilterLink *inlink){    AVFilterContext *ctx = inlink->dst;    AudioCrossoverContext *s = ctx->priv;    int sample_rate = inlink->sample_rate;    double q[16];     s->order = (s->order_opt + 1) * 2;    s->filter_count = s->order / 2;    s->first_order = s->filter_count & 1;    s->ap_filter_count = s->filter_count / 2 + s->first_order;    calc_q_factors(s->order, q);     for (int band = 0; band <= s->nb_splits; band++) {        if (s->first_order) {            set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate);            set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate);        }         for (int n = s->first_order; n < s->filter_count; n++) {            const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;             set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate);            set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate);        }         if (s->first_order)            set_ap1(&s->ap[band][0], s->splits[band], sample_rate);         for (int n = s->first_order; n < s->ap_filter_count; n++) {            const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1); 

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