/* * 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(AVFilterContext *ctx) { 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 = ff_set_common_all_channel_counts(ctx); if (ret < 0) return 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_list(ctx, sample_fmts_list); if (ret < 0) return ret; return ff_set_common_all_samplerates(ctx); } 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); set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate); } } switch (inlink->format) { case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break; case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break; default: return AVERROR_BUG; } s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 + ctx->nb_outputs * ctx->nb_outputs * 10)); if (!s->xover) return AVERROR(ENOMEM); return 0; } static int filter_frame(AVFilterLink *inlink, AVFrame *in) { AVFilterContext *ctx = inlink->dst; AudioCrossoverContext *s = ctx->priv; AVFrame **frames = s->frames; int ret = 0; for (int i = 0; i < ctx->nb_outputs; i++) { frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples); if (!frames[i]) { ret = AVERROR(ENOMEM); break; } frames[i]->pts = in->pts; } if (ret < 0) goto fail; ff_filter_execute(ctx, s->filter_channels, in, NULL, FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); for (int i = 0; i < ctx->nb_outputs; i++) { if (ff_outlink_get_status(ctx->outputs[i])) { av_frame_free(&frames[i]); continue; } ret = ff_filter_frame(ctx->outputs[i], frames[i]); frames[i] = NULL; if (ret < 0) break; } fail: for (int i = 0; i < ctx->nb_outputs; i++) av_frame_free(&frames[i]); return ret; } static int activate(AVFilterContext *ctx) { AVFilterLink *inlink = ctx->inputs[0]; int status, ret; AVFrame *in; int64_t pts; for (int i = 0; i < ctx->nb_outputs; i++) { FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[i], ctx); } ret = ff_inlink_consume_frame(inlink, &in); if (ret < 0) return ret; if (ret > 0) { ret = filter_frame(inlink, in); av_frame_free(&in); if (ret < 0) return ret; } if (ff_inlink_acknowledge_status(inlink, &status, &pts)) { for (int i = 0; i < ctx->nb_outputs; i++) { if (ff_outlink_get_status(ctx->outputs[i])) continue; ff_outlink_set_status(ctx->outputs[i], status, pts); } return 0; } for (int i = 0; i < ctx->nb_outputs; i++) { if (ff_outlink_get_status(ctx->outputs[i])) continue; if (ff_outlink_frame_wanted(ctx->outputs[i])) { ff_inlink_request_frame(inlink); return 0; } } return FFERROR_NOT_READY; } static av_cold void uninit(AVFilterContext *ctx) { AudioCrossoverContext *s = ctx->priv; av_freep(&s->fdsp); av_frame_free(&s->xover); } static const AVFilterPad inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, }, }; const AVFilter ff_af_acrossover = { .name = "acrossover", .description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."), .priv_size = sizeof(AudioCrossoverContext), .priv_class = &acrossover_class, .init = init, .activate = activate, .uninit = uninit, FILTER_INPUTS(inputs), .outputs = NULL, FILTER_QUERY_FUNC(query_formats), .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS | AVFILTER_FLAG_SLICE_THREADS, };